Re: [asterisk-users] SJphone behind NAT/Firewall without sound

[Amit Nagpal]

Is the Asterisk server yours? I am trying to figure out if Asterisk is in
your control and if it could be a problem at Asterisk, rather than your
SJPhone or your script, because I don't see any glaring problems in the
script.

Regards,
Amit.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kazabe
Sent: Friday, April 04, 2008 9:00 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SJphone behind NAT/Firewall without sound

Hi.

I need connect some LAN stations with SJphone to an Asterisk Server
published on Internet.

My Lan Clients access to Internet using a small linux firewall/proxy
server.  I use the next firewall script.  That is a simple script with
default policy ACCEPT, and NAT to share Internet.I can connect to
the asterisk server, authtenticate the users in the server, and dial
to any extension,  but we can ear any sound.I need some additional
rules in my script?

Thanks in advance

#!/bin/bash
IPTABLES=/sbin/iptables
EXT="eth0"
INT="eth1"
case "$1" in
start)
echo "1" > /proc/sys/net/ipv4/ip_forward
$IPTABLES -F INPUT
$IPTABLES -F OUTPUT
$IPTABLES -F FORWARD
$IPTABLES -F
$IPTABLES -t nat -F
$IPTABLES -t nat -A POSTROUTING -s 192.168.12.0/24 -d
0.0.0.0/0 -o
$EXT -j MASQUERADE
$IPTABLES -t nat -A PREROUTING -p TCP -s 192.168.12.0/24
--dport 80
-d -j REDIRECT --to-port 3128
$IPTABLES -A INPUT -i $EXT -p ICMP -j ACCEPT
$IPTABLES -A INPUT -i $EXT -p TCP --dport 22 -m state
--state NEW -j ACCEPT
$IPTABLES -A INPUT -i $EXT -p TCP --dport 443 -m state
--state NEW -j ACCEPT
$IPTABLES -A INPUT -i $EXT -p TCP --dport 80 -m state
--state NEW -j ACCEPT
$IPTABLES -A INPUT -p TCP -m state --state RELATED -j ACCEPT
$IPTABLES -A INPUT -i $EXT -m state --state NEW,INVALID -j
DROP
$IPTABLES -A FORWARD -i $EXT -m state --state NEW,INVALID -j
DROP
;;
stop)
$IPTABLES -F INPUT
$IPTABLES -F OUTPUT
$IPTABLES -F FORWARD
$IPTABLES -F
$IPTABLES -t nat -F
;;
restart)
$0 stop
sleep 2
$0 start
;;
status)
$IPTABLES -L
$IPTABLES --table nat --list --exact --verbose --numeric
--line-numbers
;;
*)
echo "Usage: $0 {start|stop|restart|status}"
exit 1
esac
exit 0

-- 
"Imagination is more important than knowlege"
A.E.

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[asterisk-users] Need help install rxfax/txfax

[Pete Kay]

Dear friends,

I am trying to get rxfax and txfax in my Asterisk 1.4.18 with no luck.
Everytime I get to the point Asterisk tries to compile the app_rxfax.c and
app_txfax.c I downloaded from agx-addons, the Asterisk make process crashes
with error.  I checked all the doc I can find from google but it looks like
many of the instructions are out-dated.
Could someone please send me a step-by-step guild in installing rxfax or
point me to one if there is any?

Thanks alot for all your kind help.

Regards,
Pete
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[asterisk-users] Forking using Openser And Asterisk

[Aadilkhan Maniyar]

Hi All,
 
I am stuck with an issue in the Openser+Asterisk Forking. 
 
In this solution we are using Openser as the Registrar. Hence it will
store all the contact bindings along with the q values for a given user,
say ua1. The current setup is such that the INVITEs are sent to Asterisk
by Openser and Asterisk sends out the INVITE.
 
Now if ua1 is registered with two different contacts having different q
values and i make a call from ua2 to ua1.
Openser will recieve the INVITE check for the multiple contacts of ua1
in the database. and send out an INVITE for the first contact. On
recieving a 486 busy it sends out an INVITE to the second contact. This
is where the problems lies.
 
Openser is sending Asterisk the second INVITE but none of the actions
specified in the Dialplan (extensions.conf) of Asterisk seem to be
executing on reciept of the second INVITE.
 
My extensions.conf looks like this:
exten => _.,1,NoOp(Incoming Call [EMAIL PROTECTED])
exten => _.,2,Dial(SIP/${EXTEN})
exten => _.,3,HangUp()
exten => h,4,HangUp()
 
 
This is the ouput at the asterisk cli:
 
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/ua2-0921a250", "Incoming Call
[EMAIL PROTECTED]") in new stack
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/ua2-0921a250", "SIP/ua1") in new
stack
-- Called ua1
[Apr  3 17:02:41] NOTICE[20198]: chan_sip.c:2918 auto_congest:
Auto-congesting SIP/ua1-0921f080
-- SIP/ua1-0921f080 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:3] Hangup("SIP/ua2-0921a250", "") in new
stack
  == Spawn extension (call, ua1, 3) exited non-zero on
'SIP/ua2-0921a250'
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/ua2-0921a250", "Incoming Call 
from
house extension  for [EMAIL PROTECTED]") in new stack
-- Executing [EMAIL PROTECTED]:3] Dial("SIP/ua2-0921a250", "SIP/h") in new
stack
[Apr  3 17:02:51] WARNING[21381]: chan_sip.c:2898 create_addr: No such
host: h
[Apr  3 17:02:51] WARNING[21381]: app_dial.c:1191 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [EMAIL PROTECTED]:3] Hangup("SIP/ua2-0921a250", "") in new 
stack
  == Spawn extension (call, h, 3) exited non-zero on 'SIP/ua2-0921a250'
If anyone has any inputs on this I would appreciate it..
 
Thanks & Regards,
Aadil
 
 
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Re: [asterisk-users] Click to call

[c . savinovich]

  You can try the click-to-call from www.videoreps.net... it is asterisk
based.  The sample provides you with an actual pc-to-pstn call... of
course calls to internal extensions are easier.

  There is click-to-call, and there is click-to-call-with-video

CS

>>>somebody knows some application web that allows me to call to my
>>>internal extensions of my asterisk, example  click to call.
>>>I was proving the click to call of this example but it doesn't work



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[asterisk-users] Advice on best operator phone (with attendant console)

[Faraz R. Khan]

One of our clients is using a Grandstream GXP2000 with an attendant
console. We have used the same phone with past clients successfully
however this particular operator processes around 200 calls a hours and
the GXP2000 for sure does not like the quick line shuffling and call
volume. We get the following problems randomly:

1. menu stops working
2. transfer key stops working
3. Line 1 LED gets stuck
4. Voice 'gaps' (blackouts) for 4-5 seconds
5. The phone also completely locks up regularly
6. ping response goes from 8ms to 3000ms (after which the phone locks
up)

Wondering which operator phone would work best. I have the following
choices:

1. Linksys SPA 932/962 with attendant console
2. Polycom 601/650 with attendant console

I cant confirm online whether the BLF functionality will work with
Asterisk 1.2.26. Is somebody using either of these phones in a high
volume environment successfully?

Thank you.

-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.111.111.320 x200
www.emergen.biz


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Re: [asterisk-users] Click to call

[Faraz R. Khan]

Make one using phpagi:

http://phpagi.sourceforge.net/


The AGI_AsteriskManager class should let you interface directly with
Asterisk Manager. It is fairly simple if you know PHP and AMI.


On Thu, 2008-04-03 at 20:39 -0600, troxlinux wrote:
> somebody knows some application web that allows me to call to my
> internal extensions of my asterisk, example  click to call.
> 
> I was proving the click to call of this example but it doesn't work
> 
> http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html
> 
> greeting
> 
> rickygm
> 
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-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.111.111.320 x200
www.emergen.biz


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Re: [asterisk-users] CentPBX mirror?

[James Finstrom]

pbxinaflash.com (source based)
Elastix.com (rpm based)
trixbox.org (rpm based)

Jonn Taylor wrote:
> I have a some setup scripts that use centos 4 or 5 and freepbx you are 
> welcome to use them.
>
> Jonn
>
> http://www.taylortelephone.com/asterisk/
>
>
> Chris Bagnall wrote:
>   
>>> CentPBX has bit the dust I believe.
>>> 
>>>   
>> Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel 
>> support for a Dell R200 (it's usually the SAS controller that causes the 
>> problem)? I've tried PBX-in-a-Flash without success, and Trixbox is rather 
>> too "customized" for what I'm after for this deployment.
>>
>> TIA.
>>
>> Regards,
>>
>> Chris
>>   
>> 
>
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> !DSPAM:1031,47f3b86e139811044710429!
>
>
>   

-- 
James Finstrom
Rhino Equipment Corp.
All Rhino products are made in America, 100% Money Back Guarantee,
and have a 5 Year warranty. Quality and Toughness built in!!
Phone: 1-800-785-7073 ~ FAX: +1 (480) 961-1826
IP: asterisk.rhinoequipment.com ~ FWD: 633686

THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY
MATERIAL and is thus for use only by the intended recipient. If you received
this in error, please contact the sender and delete the email and its
attachments from all computers. 



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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface andPBX Appliances

[John Signorello]

My comments were in no way an indictment of "free" software.
Yes,, we use Asterisk and it is fantastic.
Yes , we use Linux.and it is fantastic.

I pay to use UltraEdit as my text editor.
I like it better than the "free" ones that are out.

Does suggesting that some proprietary software has
features whose benefits outweigh some of the "free" ones
mean I an denigrating all "free" software??

The answer is NO.

Do you load every distribution of Linux on your machine?
No, you use the distribution that has the features you like and need.
If you had to pay $10 for your favorite distro, would you stop using it?
Probably not.
If you had to pay $100 for your distro, would you stop using it?
Hard to say, you would probably weigh the relative benefits of the $100 
distro versus
the free ones. If the $100 distro had features, whose benefits (to you) 
exceeded those of the free ones,
you might buy it. Is that analysis an indictment of all "free" software?

Of course not.





Kristian Kielhofner wrote:
> On 4/3/08, John Signorello <[EMAIL PROTECTED]> wrote:
>   
>>  John:
>>
>>  CogoBlue is a proprietary software package written by ISPBX.
>>  It is not open source. It is currently only available on ISPBX hardware.
>> 
>
> ISPBX hardware uses Asterisk, probably Linux, and probably dozens (if
> not more) FOSS applications, libraries, etc.
>
>   
>>  Check out CogoBlue, once you see what a configuration package should be,
>>  you may have to reassess what  that "free" software is really costing you.
>>
>> 
>
>   You're new here.
>
>   This is extremely offensive.  "Free" software gave you and your
> company a product (ISPBX) and a market (CogoBlue).  Where would you be
> without the "free" software projects (Asterisk, Linux, etc) ispbx
> uses?  Where would you be without the Asterisk community (hint - you
> wouldn't have a market for CogoBlue).  I'm usually not one to feed the
> trolls but this comment is over the top.
>
>   


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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface andPBX Appliances

[Kristian Kielhofner]

On 4/3/08, John Signorello <[EMAIL PROTECTED]> wrote:
>
>  John:
>
>  CogoBlue is a proprietary software package written by ISPBX.
>  It is not open source. It is currently only available on ISPBX hardware.

ISPBX hardware uses Asterisk, probably Linux, and probably dozens (if
not more) FOSS applications, libraries, etc.

>  Check out CogoBlue, once you see what a configuration package should be,
>  you may have to reassess what  that "free" software is really costing you.
>

  You're new here.

  This is extremely offensive.  "Free" software gave you and your
company a product (ISPBX) and a market (CogoBlue).  Where would you be
without the "free" software projects (Asterisk, Linux, etc) ispbx
uses?  Where would you be without the Asterisk community (hint - you
wouldn't have a market for CogoBlue).  I'm usually not one to feed the
trolls but this comment is over the top.

-- 
Kristian Kielhofner

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[asterisk-users] SJphone behind NAT/Firewall without sound

[kazabe]

Hi.

I need connect some LAN stations with SJphone to an Asterisk Server
published on Internet.

My Lan Clients access to Internet using a small linux firewall/proxy
server.  I use the next firewall script.  That is a simple script with
default policy ACCEPT, and NAT to share Internet.I can connect to
the asterisk server, authtenticate the users in the server, and dial
to any extension,  but we can ear any sound.I need some additional
rules in my script?

Thanks in advance

#!/bin/bash
IPTABLES=/sbin/iptables
EXT="eth0"
INT="eth1"
case "$1" in
start)
echo "1" > /proc/sys/net/ipv4/ip_forward
$IPTABLES -F INPUT
$IPTABLES -F OUTPUT
$IPTABLES -F FORWARD
$IPTABLES -F
$IPTABLES -t nat -F
$IPTABLES -t nat -A POSTROUTING -s 192.168.12.0/24 -d 0.0.0.0/0 
-o
$EXT -j MASQUERADE
$IPTABLES -t nat -A PREROUTING -p TCP -s 192.168.12.0/24 
--dport 80
-d -j REDIRECT --to-port 3128
$IPTABLES -A INPUT -i $EXT -p ICMP -j ACCEPT
$IPTABLES -A INPUT -i $EXT -p TCP --dport 22 -m state --state 
NEW -j ACCEPT
$IPTABLES -A INPUT -i $EXT -p TCP --dport 443 -m state --state 
NEW -j ACCEPT
$IPTABLES -A INPUT -i $EXT -p TCP --dport 80 -m state --state 
NEW -j ACCEPT
$IPTABLES -A INPUT -p TCP -m state --state RELATED -j ACCEPT
$IPTABLES -A INPUT -i $EXT -m state --state NEW,INVALID -j DROP
$IPTABLES -A FORWARD -i $EXT -m state --state NEW,INVALID -j 
DROP
;;
stop)
$IPTABLES -F INPUT
$IPTABLES -F OUTPUT
$IPTABLES -F FORWARD
$IPTABLES -F
$IPTABLES -t nat -F
;;
restart)
$0 stop
sleep 2
$0 start
;;
status)
$IPTABLES -L
$IPTABLES --table nat --list --exact --verbose --numeric 
--line-numbers
;;
*)
echo "Usage: $0 {start|stop|restart|status}"
exit 1
esac
exit 0

-- 
"Imagination is more important than knowlege"
A.E.

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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface andPBX Appliances

[John Signorello]

John:

CogoBlue is a proprietary software package written by ISPBX.
It is not open source. It is currently only available on ISPBX hardware.

Check out CogoBlue, once you see what a configuration package should be,
you may have to reassess what  that "free" software is really costing you.

The on-line documentation is extensive. There are movies that
show you the product in use.


John Faubion wrote:
>> As far as a license is concerned, we do not ship with any 
>> codecs that require licensing (we support them) and when 
>> someone purchases an ISPBX PBX system, the license for using 
>> 
>
> What Kristian was asking is, what license does the software you have written
> use? Is it GPL? Seeing that many of us only run open source software due to
> being burned by proprietary software and systems, Cogoblue would be better
> received if it were open source.
>
> John
>
>
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[asterisk-users] Click to call

[troxlinux]

somebody knows some application web that allows me to call to my
internal extensions of my asterisk, example  click to call.

I was proving the click to call of this example but it doesn't work

http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html

greeting

rickygm

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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[Matt Watson]

I've takena  quick peak at it before... but I don;t know anybody that has 
actually used it... I do intend on giving it a try myself though... it comes 
with a very very basic sample.

--
Matt


From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez [EMAIL 
PROTECTED]
Sent: Thursday, April 03, 2008 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

Matt Watson escribió:
> There is a .NET 1.1 library out there... I've played with it a little bit, 
> but not enough that I could comment on how feature rich or stable it is...
>
> http://www.voip-info.org/wiki/view/Asterisk+.NET
>
> It'll more than likely not be compatible with AMI 1.1 however, which I 
> believe is included in ast 1.6
>
> --
> Matt
>
>
Do you, or someone else,  know where to get some example about using it?

Thank you

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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface andPBX Appliances

[John Faubion]

> As far as a license is concerned, we do not ship with any 
> codecs that require licensing (we support them) and when 
> someone purchases an ISPBX PBX system, the license for using 

What Kristian was asking is, what license does the software you have written
use? Is it GPL? Seeing that many of us only run open source software due to
being burned by proprietary software and systems, Cogoblue would be better
received if it were open source.

John


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Re: [asterisk-users] Web page to show online extensions?

[Olivier]

I can't really see a business case where an operator needs to monitor a
large number of extensions : if a switch attendant has a lot of extensions
to manage, (s)he won't spend a lot of time on each incoming call.
Chances are (s)he will just transfer incoming calls to destinees leaving
screening or queues features forward those calls to secretaries if bosses
can't be disturbed.

IMHO, those secretaries are the end users of FOP, not switch attendants.
And FOP is perfect for secretaries ...
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Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out

[Mojo with Horan & Company, LLC]

sean darcy wrote:
> Kevin P. Fleming wrote:
>   
>> Mojo with Horan & Company, LLC wrote:
>>
>> 
>>> P.S.  If you can't dial seven digit numbers in your area, but you miss 
>>> it, you can restore that behavior if you feel like selecting a default 
>>> area code:
>>>
>>> exten => _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
>>>
>>> Here, if I dial a seven digit number, asterisk dials 907 followed by my 
>>> seven digits out the phone line.
>>>   
>> Well, sort of. This will also trigger if you dial the first 7 digits of
>> a 10-digit number from a device that doesn't dial 'en bloc', since there
>> is no longer any way to distinguish 7-vs-10 digit numbers by the number
>> pattern. In other words, this will work fine if you are dialing from a
>> SIP phone, but not if you are dialing from an analog phone.
>>
>> 
>
> With some trepidation, I can say my home system doesn't seem to work 
> that way. Using an analog phone, I can deal 3, 7, 10 or 11 numbers and 
> all goes as I expect.
>
> After seeing this post, I wondered why :). It seems * waits about 4 secs 
> to see if all the numbers are dialed. Or is it some fortuitous order of 
> the includes ( vaguely remembering posts about how extensions were 
> searched)?
>
> extensions.conf:
> [internal]
> include => outbound-local
> include => outbound-long-distance
> include => office-extensions
>
> [outbound-local]
> exten => _NXX,1,Answer()
> exten => _NXX,n,Dial(${faxline}/${EXTEN})
>
> [outbound-long-distance]
> exten =>_1NXXNXX,1,Answer()
> exten =>_1NXXNXX,n,Dial(iax2/office/${EXTEN})
>
> exten =>_NXXNXX,1,Answer()
> exten =>_NXXNXX,n,Dial(iax2/office/${EXTEN})
>
> [office-extensions]
> exten =>_1XX,1,Answer()
> exten =>_1XX,n,Dial(iax2/office/${EXTEN})
>
>
>
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>   

I was led to believe that it WOULD wait a few seconds, unless the '!' 
match character was on there in the dialplan.  Kevin led me to believe 
otherwise though.  Any further input, anyone?
Moj

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Re: [asterisk-users] Sending audio to a channel

[John]

I will see what I can find I just joined the list today.

--John

-Original Message-
From: Mojo with Horan & Company, LLC [mailto:[EMAIL PROTECTED]
Sent: Thu 4/3/2008 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sending audio to a channel
 
On 3/25 Justin Newman wrote a message to the list mentioning his 
SystemAnnounce application that broadcasts audio to all active channels, 
I suspect his code would be easy to modify to broadcast to a single 
channel...

Moj

John Hass wrote:
> I have a voicemail application that users can listen to messages and
> leave messages.  I am looking for a way to play a beep tone to a user
> when a new message is received when they are on the phone.
>
> Here is what I have come up with:
>
> in extensions.conf:
> [beepvoicemail]
> exten => 1000,1,answer()
> exten => 1000,2,NoCDR()
> exten => 1000,3,wait(2)
> exten => 1000,4,Set(TIMEOUT(absolute)=5)
> exten => 1000,5,playback(voicemail/beeps)
> exten => 1000,7,SendDTMF(9)
> exten => 1000,8,hangup()
>
> exten => 2000,1,Set(TIMEOUT(absolute)=5)
> exten => 2000,2,NoCDR()
> exten => 2000,3,extenspy(,g(${mailbox})WqX)
> exten => 2000,4,hangup()
>
>
> Here is what I run:
> Action: Originate
> Channel: Local/[EMAIL PROTECTED]
> MaxRetries: 0
> RetryTime: 15
> Context: beepvoicemail
> Exten: 1000
> Priority: 1
> Callerid: Pager <1000>
> Variable: mailbox=$mailbox_user
>
> I am using perl to originate so lets say mailbox 80085 left a message
> for 8675309 $mailbox_user would contain 8675309 everyone that is logged
> onto the system is part of there own spygroup the spygroup is always the
> mailbox number.
>
> This works when it doesn't crash Asterisk or the application does not
> get stuck on extenspy for hours and hours.
>
> Is there anyway to have an application that can just send audio to a
> channel without having to use extenspy (it's sort of overkill for what I
> need)
>
> Thanks For the help.
>
> --John
>
>
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Re: [asterisk-users] Sending audio to a channel

[Mojo with Horan & Company, LLC]

On 3/25 Justin Newman wrote a message to the list mentioning his 
SystemAnnounce application that broadcasts audio to all active channels, 
I suspect his code would be easy to modify to broadcast to a single 
channel...

Moj

John Hass wrote:
> I have a voicemail application that users can listen to messages and
> leave messages.  I am looking for a way to play a beep tone to a user
> when a new message is received when they are on the phone.
>
> Here is what I have come up with:
>
> in extensions.conf:
> [beepvoicemail]
> exten => 1000,1,answer()
> exten => 1000,2,NoCDR()
> exten => 1000,3,wait(2)
> exten => 1000,4,Set(TIMEOUT(absolute)=5)
> exten => 1000,5,playback(voicemail/beeps)
> exten => 1000,7,SendDTMF(9)
> exten => 1000,8,hangup()
>
> exten => 2000,1,Set(TIMEOUT(absolute)=5)
> exten => 2000,2,NoCDR()
> exten => 2000,3,extenspy(,g(${mailbox})WqX)
> exten => 2000,4,hangup()
>
>
> Here is what I run:
> Action: Originate
> Channel: Local/[EMAIL PROTECTED]
> MaxRetries: 0
> RetryTime: 15
> Context: beepvoicemail
> Exten: 1000
> Priority: 1
> Callerid: Pager <1000>
> Variable: mailbox=$mailbox_user
>
> I am using perl to originate so lets say mailbox 80085 left a message
> for 8675309 $mailbox_user would contain 8675309 everyone that is logged
> onto the system is part of there own spygroup the spygroup is always the
> mailbox number.
>
> This works when it doesn't crash Asterisk or the application does not
> get stuck on extenspy for hours and hours.
>
> Is there anyway to have an application that can just send audio to a
> channel without having to use extenspy (it's sort of overkill for what I
> need)
>
> Thanks For the help.
>
> --John
>
>
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Web page to show online extensions?

[Al lists]

you can see users status in Jaber,
Install Open fire Jabber server with Asterisk pluging.


On Thu, Apr 3, 2008 at 1:55 PM, Earl Terwilliger <[EMAIL PROTECTED]> wrote:

> On Thursday 03 April 2008 02:59:07 pm faraz wrote:
> > FOP is quite clunky!
>
> one reason i wrote the event montor... which is in PHP (and Ajax or rather
> Ajap) and does not poll the asterisk manager (which in my opinion
> overloads
> asterisk)
>
> >
> > Also the flash is almost un-usable with a large number of extensions
> > Would love to see something in PHP/Ajax which could be lightweight and
> > fast.
> >
> > We are working on something along those lines which we should be able to
> > release in a few months.
> >
> > On Thu, 2008-04-03 at 14:42 -0400, Dean Collins wrote:
> > > Cute :)
> > >
> > > I was thinking about getting something more complex developed but yes
> > > FOP is a great product though getting a little old.time for the
> > > next
> > > version?
> > >
> > >
> > >
> > > Regards,
> > >
> > > Dean Collins
> > > Cognation Pty Ltd
> > > [EMAIL PROTECTED]
> > > +1-212-203-4357
> > > +61-2-9016-5642 (Sydney in-dial).
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > >
> > > [mailto:asterisk-users-
> > >
> > > > [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> > > > Sent: Thursday, 3 April 2008 2:32 PM
> > > > To: asterisk-users@lists.digium.com
> > > > Subject: Re: [asterisk-users] Web page to show online extensions?
> > > >
> > > > On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
> > > > > What about building an Adobe AIR application that can do this.
> > > >
> > > > Any application that connects to the manager interface can do that.
> > >
> > > It
> > >
> > > > can be AIR, or FIRE or GROUND.
> > > >
> > > > The FOP exists and does that.
> > > >
> > > > --
> > > >Tzafrir Cohen
> > > > icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> > > > +972-50-7952406   mailto:[EMAIL PROTECTED]
> > > > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> > > >
> > > > ___
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> > > --
> > >
> > > > asterisk-users mailing list
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> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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Re: [asterisk-users] Web page to show online extensions?

[Mojo with Horan & Company, LLC]

faraz wrote:
> FOP is quite clunky! 
>
> Also the flash is almost un-usable with a large number of extensions
> Would love to see something in PHP/Ajax which could be lightweight and
> fast.
>   
Last version of FOP I downloaded had a DHTML client in addition to the 
fat Flash client, I'm pretty happy with that.  I embed it into our 
windows boxen's desktops, works great!

Moj

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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[sanjay . rajdev]

Thanks a lot, will try this out. 

Regards,
Sanjay.

- Original Message -
From: "Grey Man" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 4, 2008 4:58:56 AM (GMT+0530) Asia/Calcutta
Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

On Fri, Apr 4, 2008 at 12:11 AM,
<[EMAIL PROTECTED]> wrote:
> Can you Please refer me to any, the one that I found are all either in 
> Java/C. Or if they are in C# they are not opensource.
>

I know www.mysipswitch is written in C# and can place SIP calls to
Asterisk servers. The code is open sourced at
http://www.codeplex.com/mysipswitch.

Regards,

Greyman.

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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Al lists]

Darren and Jerry,
is it possible to post your config if its different than:
http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check

thank you!


On Thu, Apr 3, 2008 at 8:18 AM, Darren Wright <[EMAIL PROTECTED]> wrote:

> I've used Adit600's almost exclusively for my installs.   All have worked
> great for me.
>
> -D
>
>
> 
>
> From: [EMAIL PROTECTED] on behalf of Steve Totaro
> Sent: Thu 4/3/2008 10:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Need some input for Quad T1 and channel
> banks.
>
>
>
> Just Google Quintum Tenor AX.  Well worth the money.
>
> Thanks,
> Steve Totaro
>
> On Mon, Mar 31, 2008 at 10:03 PM, Al lists <[EMAIL PROTECTED]> wrote:
> > Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
> >  Steve, what are my options for SIP to fxs?
> >  thank you!
> >
> >
> >
> >  On 3/31/08, Doug Lytle <[EMAIL PROTECTED]> wrote:
> >  > Don Pobanz wrote:
> >  > > Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
> >  > >
> >  > >>
> >  > >
> >  > > This does not sound right. If it is 2 PRIs then it should be 46
> channels
> >  > >
> >  > >
> >  >
> >  > I may have the terminology incorrect. I don't have a D channel, so I
> >  > guess this would be called a T1 then?
> >  >
> >  > Doug
> >  >
> >  >
> >  > --
> >  > Ben Franklin quote:
> >  >
> >  > "Those who would give up Essential Liberty to purchase a little
> Temporary
> >  > Safety, deserve neither Liberty nor Safety."
> >  >
> >  >
> >  > ___
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> http://www.api-digital.com/>  --
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> >  >
> >
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>
>
> This message was sent from D2 Technology, INC.
>
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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[Grey Man]

On Fri, Apr 4, 2008 at 12:11 AM,
<[EMAIL PROTECTED]> wrote:
> Can you Please refer me to any, the one that I found are all either in 
> Java/C. Or if they are in C# they are not opensource.
>

I know www.mysipswitch is written in C# and can place SIP calls to
Asterisk servers. The code is open sourced at
http://www.codeplex.com/mysipswitch.

Regards,

Greyman.

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[asterisk-users] About outdail SIPCALLID

[Mike Wang]

Hi
 I sent this 3 hours ago, seems not go through, so sent again.

 I have an asterisk php-agi application.
 It answer's call , then outdial to another number:

$agi->exec_dial("SIP", [EMAIL PROTECTED] , "20", $options);

  How can I get a SIPCALLID for this out-dialed call?

   The SIPCALLID seems the incoming call's SIPCALLID.



Thanks.

Mike

-- 
Best Regards

Mike

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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[sanjay . rajdev]

Can you Please refer me to any, the one that I found are all either in Java/C. 
Or if they are in C# they are not opensource.

Regards,
Sanjay.

- Original Message -
From: "Grey Man" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 4, 2008 4:32:44 AM (GMT+0530) Asia/Calcutta
Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

> On Thu, Apr 3, 2008 at 11:35 PM,  <[EMAIL PROTECTED]> wrote:
>  Do anyone has an idea about an open source SIP API written in C# that can 
> communicate with Asterisk, to call out?

There are a few C# SIP stacks around that will let you do that.
Creating a call from such a stack to Asterisk will be the same as to
any other SIP server.

Regards,

Greyman.

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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[Grey Man]

> On Thu, Apr 3, 2008 at 11:35 PM,  <[EMAIL PROTECTED]> wrote:
>  Do anyone has an idea about an open source SIP API written in C# that can 
> communicate with Asterisk, to call out?

There are a few C# SIP stacks around that will let you do that.
Creating a call from such a stack to Asterisk will be the same as to
any other SIP server.

Regards,

Greyman.

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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[Rodrigo Gonzalez]

Matt Watson escribió:

There is a .NET 1.1 library out there... I've played with it a little bit, but 
not enough that I could comment on how feature rich or stable it is...

http://www.voip-info.org/wiki/view/Asterisk+.NET

It'll more than likely not be compatible with AMI 1.1 however, which I believe 
is included in ast 1.6

--
Matt

  

Do you, or someone else,  know where to get some example about using it?

Thank you



smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out

[sean darcy]

Kevin P. Fleming wrote:
> Mojo with Horan & Company, LLC wrote:
> 
>> P.S.  If you can't dial seven digit numbers in your area, but you miss 
>> it, you can restore that behavior if you feel like selecting a default 
>> area code:
>>
>> exten => _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
>>
>> Here, if I dial a seven digit number, asterisk dials 907 followed by my 
>> seven digits out the phone line.
> 
> Well, sort of. This will also trigger if you dial the first 7 digits of
> a 10-digit number from a device that doesn't dial 'en bloc', since there
> is no longer any way to distinguish 7-vs-10 digit numbers by the number
> pattern. In other words, this will work fine if you are dialing from a
> SIP phone, but not if you are dialing from an analog phone.
> 

With some trepidation, I can say my home system doesn't seem to work 
that way. Using an analog phone, I can deal 3, 7, 10 or 11 numbers and 
all goes as I expect.

After seeing this post, I wondered why :). It seems * waits about 4 secs 
to see if all the numbers are dialed. Or is it some fortuitous order of 
the includes ( vaguely remembering posts about how extensions were 
searched)?

extensions.conf:
[internal]
include => outbound-local
include => outbound-long-distance
include => office-extensions

[outbound-local]
exten => _NXX,1,Answer()
exten => _NXX,n,Dial(${faxline}/${EXTEN})

[outbound-long-distance]
exten =>_1NXXNXX,1,Answer()
exten =>_1NXXNXX,n,Dial(iax2/office/${EXTEN})

exten =>_NXXNXX,1,Answer()
exten =>_NXXNXX,n,Dial(iax2/office/${EXTEN})

[office-extensions]
exten =>_1XX,1,Answer()
exten =>_1XX,n,Dial(iax2/office/${EXTEN})



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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[sanjay . rajdev]

This work with Asterisk Manager Interface. I want to implement basic phone 
functionality in C#.

Regards,
Sanjay.

- Original Message -
From: "Matt Watson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, April 4, 2008 3:48:52 AM (GMT+0530) Asia/Calcutta
Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

There is a .NET 1.1 library out there... I've played with it a little bit, but 
not enough that I could comment on how feature rich or stable it is...

http://www.voip-info.org/wiki/view/Asterisk+.NET

It'll more than likely not be compatible with AMI 1.1 however, which I believe 
is included in ast 1.6

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, April 03, 2008 5:28 PM
To: asterisk-users
Subject: [asterisk-users] C# SIP API to Comiunicate with Asterisk

Do anyone has an idea about an open source SIP API written in C# that can 
communicate with Asterisk, to call out?

Regards,
Sanjay.


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Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

[Matt Watson]

There is a .NET 1.1 library out there... I've played with it a little bit, but 
not enough that I could comment on how feature rich or stable it is...

http://www.voip-info.org/wiki/view/Asterisk+.NET

It'll more than likely not be compatible with AMI 1.1 however, which I believe 
is included in ast 1.6

--
Matt

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, April 03, 2008 5:28 PM
To: asterisk-users
Subject: [asterisk-users] C# SIP API to Comiunicate with Asterisk

Do anyone has an idea about an open source SIP API written in C# that can 
communicate with Asterisk, to call out?

Regards,
Sanjay.


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[asterisk-users] Hearing "transfer" during call

[Vincent Li]

Hi list,

I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word "transfer", I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:

sip.conf:

[ht286]
type=friend
regexten=6010
username=ht286
secret=secret
context=numberplan-local
callerid="Home Phone" <6010>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
[EMAIL PROTECTED]
dtmfmode=rfc2833

extensions.conf:

[macro-stdexten]
exten => s,1,Dial(${ARG2},20,t)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${ARG1},u)
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(${ARG1},b)
exten => s-BUSY,2,Goto(default,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)
exten => a,1,VoicemailMain(${ARG1})

[default]
exten => s,1,Ringing
exten => s,n,Wait(1)
exten => s,n,Answer
exten => s,n,Wait(1)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,Background(if-u-know-ext-dial)
exten => s,n,Background(otherwise)
exten => s,n,Background(to-reach-operator)
exten => s,n,Background(pls-hold-while-try)
exten => s,n,WaitExten(6)
exten => s,n,Hangup()
exten => i,1,Playback(invalid)
exten => i,n,Goto(s,1)
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()

include => internal


[internal]

; define local extensions here

exten => 6010,1,Macro(stdexten,${EXTEN},SIP/ht286)

[numberplan-local]
ignorepat => 9
include => default
include => parkedcalls
comment => Local Calling

include => internal

features.conf:

[general]
parkext => 700  ; What ext. to dial to park
parkpos => 701-720  ; What extensions to park calls on
context => parkedcalls  ; Which context parked calls are in
;context => numberplan-local; Which context parked calls are in
;parkingtime => 45  ; Number of seconds a call can be parked for
; (default is 45 seconds)
;transferdigittimeout => 3  ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep; Sound file to play to the parked caller
; when someone dials a parked call
; or the Touch Monitor is activated/deactivated.
xfersound = beep; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
;adsipark = yes ; if you want ADSI parking announcements
;findslot => next   ; Continue to the 'next' parking
space. Defaults to 'first' available
;pickupexten = *8   ; Configure the pickup extension.  Default is *8
;featuredigittimeout = 500  ; Max time (ms) between digits for
; feature activation.  Default is 500


[featuremap]
blindxfer => #  ; Blind transfer
;disconnect => *0   ; Disconnect
;automon => *1  ; One Touch Record (a.k.a. Touch Monitor)
atxfer => * ; A

users.conf:

[6004]
fullname = Analog User 4
secret = 6004
email =
cid_number =
zapchan = 4
context = numberplan-local
hasvoicemail = yes
hasdirectory = yes
hassip = no
hasiax = no
hasmanager = no
callwaiting = no
threewaycalling = no
mailbox = 6004
hasagent = no
group = 2

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[asterisk-users] Vitelity and AsteriskNOW

[Roderick A. Anderson]

I wanted to try AsteriskNOW plus a few others to see which I can wrap my 
head around the quickest.

The issue so far is I can't figure out how to use my Vitelity account 
with it.  I went so far as to put their Asterisk configuration in the 
sip.conf file but still no joy.

Any pointer as to where to search?  I found a few threads in the 
AsteriskNOW forums and one thread from last year on this list but 
nothing on configuration.  Just reliability and service discussions.

I filed a trouble ticket with Vitelity and I'll head over to the 
AsteriskNOW forums but they seem pretty quiet.


Rod
-- 

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[asterisk-users] C# SIP API to Comiunicate with Asterisk

[sanjay . rajdev]

Do anyone has an idea about an open source SIP API written in C# that can 
communicate with Asterisk, to call out?

Regards,
Sanjay.


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Re: [asterisk-users] IVR Asterisk Voice Recognition -Asterisk withlumenvox

[Al Baker]

Quote "Re-reading your answer it appears you provided advice about a topic
where you hadn't actually utilized the product."

No - I asked a question  to find out peoples experience.
I did not Offer advice to anybody, I ASKED for advice vis-a-vis where They had 
actually
used the product.


And yes - if anyone wishes to share more about their experiences I'd like to 
hear it.



Dean Collins wrote:
> Ok, I think that original request was answered more than a few days ago
> by people who had actually used Lumenvox.
>
> Re-reading your answer it appears you provided advice about a topic
> where you hadn't actually utilized the product.
>
> As several people who answered Phillip's original question - Lumenvox is
> a good product and at an affordable price point.
>
> Just understand it is utterance recognition with limited response set
> recognition - NOT NLVR.
>
>  
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED] 
> +1-212-203-4357
> +61-2-9016-5642 (Sydney in-dial). 
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Al Baker
>> Sent: Thursday, 3 April 2008 12:14 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] IVR Asterisk Voice Recognition -Asterisk
>> withlumenvox
>>
>> Thank you.
>> I think that part of one's research is to ask in a user group what
>> peoples "real word" experience of a product has been. Particularly
>> 
> since
>   
>> I was responding to a post by Philip in which he had said
>>   "So what is that you'd like to know?
>>Philipp"
>>
>> in response to  a question of
>>
>>"I would like to know from someone uses or has used the engines of
>>LumenVox for integration with the asterisk for voice recognition."
>>
>> posted by another member of the mailing list.
>>
>> Sorry you don't agree.
>>
>> However, I would still be interested in hearing anyone experiences
>> 
> that would care
>   
>> to share them.
>>
>> And to those kind enough to share I say 'thanks' in advance.
>>
>> g
>> Dean Collins wrote:
>> 
>>> Hi Al,
>>>
>>> I'm saying this politely so don't take it the wrong way. Go away and
>>>   
> do
>   
>>> some research.
>>>
>>> Learn the difference between NLVR (Dragon Dictate) and limited set
>>> utterance recognition (Lumenvox).
>>>
>>> Lumenvox is a great product for the price point and as their
>>>   
> developer
>   
>>> kit is so reasonable you should buy one to have a play on a test
>>>   
> bench
>   
>>> with as a basic minimum.
>>>
>>>
>>>
>>> Regards,
>>>
>>> Dean Collins
>>> Cognation Pty Ltd
>>> [EMAIL PROTECTED]
>>> +1-212-203-4357
>>> +61-2-9016-5642 (Sydney in-dial).
>>>
>>>
>>>
>>>   
 -Original Message-
 From: [EMAIL PROTECTED]
 
> [mailto:asterisk-users-
>   
 [EMAIL PROTECTED] On Behalf Of Al Baker
 Sent: Wednesday, 2 April 2008 11:25 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk

 
>>> withlumenvox
>>>
>>>   
 In the real world, just how good of recognition can you get based
 
> on
>   
 your experience ?

 How much processing power do you find it takes ? I know that a

 
>>> dedicated
>>>
>>>   
 Voice Recognition for the
 PC such as "Dragon Naturally Speaking" requires :
 -a pretty beefy system
 -that you use a limited set of microphones
 - and ideally, a 20 session to "train" the software.

 Clearly all of this not feasible in a IVR environment, so, in the
 absence of all this, just how good , and how sophisticated of a
 
> voice
>   
 recognition can one achieve ?

 Philipp von Klitzing wrote:

 
> Hi!
>
>
>
>   
>> I would like to know from someone uses or has used the engines of
>> LumenVox for integration with the asterisk for voice recognition.
>>
>>
>> 
> So what is that you'd like to know?
>
> Philipp
>
>
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>   
>>> --
>>>
>>>   
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>   
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>   
 asterisk-users mailing list
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>   
>>> asterisk-users mailing list

Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Francesco Peeters (Linux)]

Drew Gibson wrote:
> 
>  I suspect that this is due to the call 
> billing structure in Europe. They make the North American telcos look 
> positively philanthropic.
>   
Yes indeed!
Flat rate calling plans? What are those?
Flat rate Mobile Internet? non-existant!

We pay per minute/SMS/MB on every plan, and the only thing you achieve
on a more expensive plan is to pay less per unit, but flat-rate is
NON-EXISTANT...

It is one of the few things I actually envy my US colleagues for! (Of
course, we do have more PTO! )

-- 
Francesco Peeters
Laptop: IBM T43 with Ubuntu Gutsy Gibbon, Workstation
Server: P4i65G, 2.4GHz with Ubuntu Gutsy Gibbon, Server Edition
   Postfix, Dovecot, Mailman, Apache2, Squirrelmail


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Re: [asterisk-users] Web page to show online extensions?

[Earl Terwilliger]

On Thursday 03 April 2008 02:59:07 pm faraz wrote:
> FOP is quite clunky!

one reason i wrote the event montor... which is in PHP (and Ajax or rather 
Ajap) and does not poll the asterisk manager (which in my opinion overloads 
asterisk)

>
> Also the flash is almost un-usable with a large number of extensions
> Would love to see something in PHP/Ajax which could be lightweight and
> fast.
>
> We are working on something along those lines which we should be able to
> release in a few months.
>
> On Thu, 2008-04-03 at 14:42 -0400, Dean Collins wrote:
> > Cute :)
> >
> > I was thinking about getting something more complex developed but yes
> > FOP is a great product though getting a little old.time for the
> > next
> > version?
> >
> >
> >
> > Regards,
> >
> > Dean Collins
> > Cognation Pty Ltd
> > [EMAIL PROTECTED]
> > +1-212-203-4357
> > +61-2-9016-5642 (Sydney in-dial).
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> >
> > [mailto:asterisk-users-
> >
> > > [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> > > Sent: Thursday, 3 April 2008 2:32 PM
> > > To: asterisk-users@lists.digium.com
> > > Subject: Re: [asterisk-users] Web page to show online extensions?
> > >
> > > On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
> > > > What about building an Adobe AIR application that can do this.
> > >
> > > Any application that connects to the manager interface can do that.
> >
> > It
> >
> > > can be AIR, or FIRE or GROUND.
> > >
> > > The FOP exists and does that.
> > >
> > > --
> > >Tzafrir Cohen
> > > icq#16849755  jabber:[EMAIL PROTECTED]
> > > +972-50-7952406   mailto:[EMAIL PROTECTED]
> > > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> > >
> > > ___
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> >
> > --
> >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users



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Re: [asterisk-users] Web page to show online extensions?

[faraz]

FOP is quite clunky! 

Also the flash is almost un-usable with a large number of extensions
Would love to see something in PHP/Ajax which could be lightweight and
fast.

We are working on something along those lines which we should be able to
release in a few months.

On Thu, 2008-04-03 at 14:42 -0400, Dean Collins wrote:
> Cute :)
> 
> I was thinking about getting something more complex developed but yes
> FOP is a great product though getting a little old.time for the
> next
> version?
> 
>  
> 
> Regards,
> 
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED] 
> +1-212-203-4357
> +61-2-9016-5642 (Sydney in-dial). 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> > Sent: Thursday, 3 April 2008 2:32 PM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Web page to show online extensions?
> > 
> > On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
> > > What about building an Adobe AIR application that can do this.
> > 
> > Any application that connects to the manager interface can do that.
> It
> > can be AIR, or FIRE or GROUND.
> > 
> > The FOP exists and does that.
> > 
> > --
> >Tzafrir Cohen
> > icq#16849755  jabber:[EMAIL PROTECTED]
> > +972-50-7952406   mailto:[EMAIL PROTECTED]
> > http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> > 
> > ___
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> --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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> 
-- 
Faraz R Khan
Chief Architect
Emergen Consulting Pvt Ltd
+92.21.111.111.320 x200
www.emergen.biz


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[asterisk-users] IAX - FWD status

[Joseph]

I've noticed that FWD updated but IAX is not registering with FWD server; nor 
the log-in page exist.
Is FWD still supporting IAX?

How to check their IAX server status? 

-- 
#Joseph

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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances

[Matt Signorello]

Kristian,

I completely pulled a my-bad and posted to the users list instead of
biz. I even had re-posted to the biz list and apologized for the
mix-up.  (This time I originally replied to your questions on the biz,
not the users.. its been a long day..)

To answer you question, we do not modify Asterisk in any way. The source
is the same as one would download directly from Digium (1.2 branch). We
wrote our own backend application and cogoblue itself completely from
scratch. We simple create the conf's copy to the Asterisk dir, and
reload.

At this time we do not use any other third party tools or applications
as part of our product. 

As far as a license is concerned, we do not ship with any codecs that
require licensing (we support them) and when someone purchases an ISPBX
PBX system, the license for using COGOBLUE is included in the purchase
price. 

We do not redistribute the Asterisk source ourselves and simply refer
people to Digium if anyone wishes to download it.

Thanks, 

Matt


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Re: [asterisk-users] AsteriskNOW and IE

[Godwin Stewart]

On Thu, 3 Apr 2008 09:39:55 + (UTC), [EMAIL PROTECTED] (Tony
Mountifield) wrote:

> nothing was shown in the main pane. So there is definitely something
> wrong with IE compatibility.

s/ compatibility//

There. I fixed your post :)

-- 
Godwin Stewart - Horwich IT services

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Re: [asterisk-users] Web page to show online extensions?

[Dean Collins]

Cute :)

I was thinking about getting something more complex developed but yes
FOP is a great product though getting a little old.time for the next
version?

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> Sent: Thursday, 3 April 2008 2:32 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Web page to show online extensions?
> 
> On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
> > What about building an Adobe AIR application that can do this.
> 
> Any application that connects to the manager interface can do that. It
> can be AIR, or FIRE or GROUND.
> 
> The FOP exists and does that.
> 
> --
>Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
> 
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Re: [asterisk-users] Web page to show online extensions?

[Lee Jenkins]

Vincent wrote:
> On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger <[EMAIL PROTECTED]>
> wrote:
>>  http://www.micpc.com/eventmonitor/
> 
> Thanks guys. I was also thinking of stand-alone apps like Jabber or
> something. The call is simply to know if an extension is on- or
> offline.
> 
> 

Not web based, but:

http://www.datatrakpos.com/pos/datatalk/maestro.aspx

-- 
Warm Regards,

Lee

"Everything I needed to learn in life, I learned selling encyclopedias door to 
door."

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Re: [asterisk-users] Web page to show online extensions?

[Tzafrir Cohen]

On Thu, Apr 03, 2008 at 07:22:55PM +0200, Vincent wrote:
> On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger <[EMAIL PROTECTED]>
> wrote:
> > http://www.micpc.com/eventmonitor/
> 
> Thanks guys. I was also thinking of stand-alone apps like Jabber or
> something. The call is simply to know if an extension is on- or
> offline.

The Druid people mentioned something about a Pidgin plugin and generally
about Asterisk as a component of Jabberd which, IIRC, exposes that
information to just about any decent XMPP client.

However, one simpler method would probably be to use SIP
publish/subscribe, which is supoprted by quite a few clients. I'm not
aware of any Pidgin / Miranda plugins. But twinkle and such can give you
a nice display of available extensions.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Web page to show online extensions?

[Tzafrir Cohen]

On Thu, Apr 03, 2008 at 01:32:37PM -0400, Dean Collins wrote:
> What about building an Adobe AIR application that can do this.

Any application that connects to the manager interface can do that. It
can be AIR, or FIRE or GROUND.

The FOP exists and does that. 

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances

[John Signorello]

No changes have been made to Asterisk.

CogoBlue is a PBX configuration tool.

It is only available at this time with ISPBX pbx appliances.


Kristian Kielhofner wrote:
> On 4/3/08, Matt Signorello <[EMAIL PROTECTED]> wrote:
>   
>> Hi Everyone,
>>
>>  My name is Matt Signorello and I'm responsible for wholesale dealers
>>  sales here at ISPBX. (www.ispbx.com)
>>
>> 
>
> Matt,
>
>   As some others have already pointed out, this list is for
> non-commercial discussion and you shouldn't have used Tony's existing
> thread to announce your product.
>
>   Anyways, I see that ispbx is "Asterisk based".  What modifications
> have you made to Asterisk?  What other tools/utilities does ispbx use?
>  What license(s) are those tools under?  Do you have a download area
> for the source (Asterisk, etc)?
>
> Thanks!
>
>   


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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances

[Kristian Kielhofner]

On 4/3/08, Matt Signorello <[EMAIL PROTECTED]> wrote:
> Hi Everyone,
>
>  My name is Matt Signorello and I'm responsible for wholesale dealers
>  sales here at ISPBX. (www.ispbx.com)
>

Matt,

  As some others have already pointed out, this list is for
non-commercial discussion and you shouldn't have used Tony's existing
thread to announce your product.

  Anyways, I see that ispbx is "Asterisk based".  What modifications
have you made to Asterisk?  What other tools/utilities does ispbx use?
 What license(s) are those tools under?  Do you have a download area
for the source (Asterisk, etc)?

Thanks!

-- 
Kristian Kielhofner

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[asterisk-users] Problems with analog <-> SIP phone confif\gurations

[Timothy Smith]

Hi,

I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on
an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,
where an incoming line is plugged and also analog phone plugged to the
FXS port. Am faced with the problems below.

- For conversations between analog phone and sip phone, Analog phone
can't here the SIP user but Sip user hears.
- Calling the PSTN from the Analog phone,  still the analog phone
can't hear but the PSTN user hears him saying "hello." repeatedly.

Any help appreciated?


I attempted a SIP debug and this is a sample out out:

<-- SIP read from 192.168.209.1:48099:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.209.253:5060;branch=z9hG4bK661b7c81;rport=5060;received=192.168.209.253
From: "asterisk" ;tag=as7b41af2a
To: "Ananth" ;tag=2bb81ff3969
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
Content-Length: 0
Server: SJphone/1.65.377a (SJ Labs)

-
--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.102.10:49166
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: 
set_destination: Parsing  for address/port
to send to
set_destination: set destination to 192.168.102.10, port 5060
Transmitting (NAT) to 192.168.209.1:48099:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.209.253:5060;branch=z9hG4bK4162221c;rport
From: "analog-phone" ;tag=as2b73e0bc
To: ;tag=2ae01fe36af
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0




Regards,
Tim

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Re: [asterisk-users] ztdummy

[Norman Franke]

On Apr 3, 2008, at 12:45 PM, [EMAIL PROTECTED]  
wrote:



You can try zttest, although I'd bet it will hang. See what's going
to the console (or use dmesg.) If it's a lot of rtc errors, then
you'll likely need to upgrade your kernel to at least 2.6.23.11. That
worked for me.


I'd be surprised if that is the solution - I have been using  
ztdummy with
the RTC hook since 2.6.9 with no problems, and again on  
2.6.18-53.1.6.el5


Unless it's a 64-bit issue - I've only ever used 32-bit.



I'm using 32-bit as well, but it may depend on the hardware. I'm  
using HP DL-380s and Debian (both etch and sarge had the same problems.)


I did nothing other than upgrade my kernel (and recompile zaptel, of  
course) and it just started to work. I used to get a lot of those  
missing interrupt errors to the console before, not now.


If zttest hangs, I'd suspect the drivers are loaded. Otherwise,  
you'll get an error that it can't find a zap device (and playback  
then works.)


Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

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[asterisk-users] NAT when outbound call leg is not a local subscriber?

[Amit Nagpal]

Hi,

I have been experimenting with NAT and Asterisk a bit now. Though I have
made progress along the way, I have come across the following problem. I'll
really appreciate if anyone can provide me any help or pointers. Thanks!

Successful Scenario:
---
All sorts of NAT calls are successful with full two-way media when both end
points are locally subscribed users. 

Problem Scenario:

UA-Local: Locally subscribed & registered user (configured in sip.conf) that
is hidden behind NAT.
UA-External: Some remote user hidden behind NAT, but registered with some
publically accessible registrar/proxy.
My Asterisk is also publically accessible (i.e. not hidden behind NAT)

When UA-Local calls out UA-External, I only get one-way audio. Specifically,
when I debugged using ethereal traces, I found that Asterisk is sending RTP
packets to the private IP of UA-External and not to the corresponding
NAT-mapped IP accessible to the outside world. So, UA-Local is able to hear
UA-External, but UE-External can't hear UA-Local. It all works perfectly
fine, if UA-External were to call UA-Local. Then I get full two-way media.
The problem is only when Asterisk calls out a non-locally subscribed user.

Brief Setup Background:
--
[EMAIL PROTECTED]: user subscribed in sip.conf
[EMAIL PROTECTED]: user subscribed in sip.conf
[EMAIL PROTECTED]: some user actively registered with some domain
external.com.

I am using OpenSER as my external proxy for external.com and I have my DNS
setup all right.

Following scenario is working fine in my setup:

UA1 <---> NAT <---> Asterisk <---> NAT <---> UA2.

Calls go through perfectly fine - with two-way media - when initiated in
either direction.

Following scenario works fine when UAE calls out UA1. But when UA1 calls out
UAE, I only get one-way audio, wherein only UA1 can hear UAE. UAE can't hear
UA1, as Asterisk keeps sending RTP packets to the private address of UAE.

UAE <--> (NAT + External-Proxy) <--> Asterisk <--> NAT <--> UA1

I am using iptable's MASQUERADE target for NAT, which by default implements
a 'Port Restricted Cone NAT' as per STUN RFC's terminology.

All my UAs are XLite-on-Windows. My Asterisk is running on Fedora Core 6.

I have the following flags set in the [general] section of my sip.conf
[general]
nat=yes
qualify=yes
rtpkeepalive=60
rtptimeout=90
rtpholdtimeout=300
canreinvite=no
context=sip_incoming
(... among others ...)

Following is the relevant portion of my extensions.conf
[sip_incoming]
exten => _.,1,GotoIf($[${SIPDOMAIN}=mydomain.com]?4)
exten => _.,2,Dial(SIP/[EMAIL PROTECTED])
exten => _.,3,HangUp()
exten => _.,4,Dial(SIP/${EXTEN})
exten => _.,5,HangUp()
exten => h,1,HangUp()

Am I doing something wrong? Or is there a bug in Asterisk, wherein, while
calling out to non-locally subscribed users, it blindly trusts the notion of
their IP address when it comes to RTP. 

Any help is highly appreciated.

Regards,
Amit.




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Re: [asterisk-users] Wait for dialtone feature on FXO device

[John Novack]

Well, it SHOULD have been obvious that IF I had degree of skill I would 
have years ago.
Those kind of comments serve no purpose other than to anger and to boost 
the already inflated ego of those who make the comments!

This is a "USERS" list after all!

John Novack

Tilghman Lesher wrote:
> On Thursday 03 April 2008 10:37:32 John Novack wrote:
>   
>> WOW! Is this LONG overdue.
>> Why this wasn't done initially is beyond me
>> It has caused so many troubles and questions and posts from folks who
>> expected Asterisk to at least have a feature that has been in a dial up
>> modem for 10+ years.
>> 
>
> Welcome to open source.  If you'd like to see a feature, might I suggest
> writing it, and contributing it back?
>
>   

-- 
Dog is my co-pilot


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[asterisk-users] Sending audio to a channel

[John Hass]

I have a voicemail application that users can listen to messages and
leave messages.  I am looking for a way to play a beep tone to a user
when a new message is received when they are on the phone.

Here is what I have come up with:

in extensions.conf:
[beepvoicemail]
exten => 1000,1,answer()
exten => 1000,2,NoCDR()
exten => 1000,3,wait(2)
exten => 1000,4,Set(TIMEOUT(absolute)=5)
exten => 1000,5,playback(voicemail/beeps)
exten => 1000,7,SendDTMF(9)
exten => 1000,8,hangup()

exten => 2000,1,Set(TIMEOUT(absolute)=5)
exten => 2000,2,NoCDR()
exten => 2000,3,extenspy(,g(${mailbox})WqX)
exten => 2000,4,hangup()


Here is what I run:
Action: Originate
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 15
Context: beepvoicemail
Exten: 1000
Priority: 1
Callerid: Pager <1000>
Variable: mailbox=$mailbox_user

I am using perl to originate so lets say mailbox 80085 left a message
for 8675309 $mailbox_user would contain 8675309 everyone that is logged
onto the system is part of there own spygroup the spygroup is always the
mailbox number.

This works when it doesn't crash Asterisk or the application does not
get stuck on extenspy for hours and hours.

Is there anyway to have an application that can just send audio to a
channel without having to use extenspy (it's sort of overkill for what I
need)

Thanks For the help.

--John


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Re: [asterisk-users] Web page to show online extensions?

[Dean Collins]

What about building an Adobe AIR application that can do this.

I'm kind of very curious about why developers haven't flocked to the AIR
platform for Asterisk apps yet.

I was looking to fund the development of an 'awareness application' for
asterisk based on AIR last year but this was dependant on Russell
Bryants 'status API development' that he was working on last year but
haven't heard/seen anything about it since then.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Vincent
> Sent: Thursday, 3 April 2008 1:23 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Web page to show online extensions?
> 
> On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger <[EMAIL PROTECTED]>
> wrote:
> > http://www.micpc.com/eventmonitor/
> 
> Thanks guys. I was also thinking of stand-alone apps like Jabber or
> something. The call is simply to know if an extension is on- or
> offline.
> 
> 
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Re: [asterisk-users] Web page to show online extensions?

[Vincent]

On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger <[EMAIL PROTECTED]>
wrote:
>   http://www.micpc.com/eventmonitor/

Thanks guys. I was also thinking of stand-alone apps like Jabber or
something. The call is simply to know if an extension is on- or
offline.


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[asterisk-users] Asterisk (or maybe Zaptel) goes to "sleep" after inactivity?

[Joshua Kinard]

Hi all,

Noticed a curious issue in my testing setup for a faxing system I'm putting 
together, but it looks like if I let the lines all sit idle for a few days (no 
one uses this yet, so the whole thing really does sit idle until I do testing 
on it or something else), something I believe on my Asterisk end goes to a kind 
of "Sleep".  It's hard to describe really, but I'm not sure if it's Asterisk 
itself or maybe the Zaptel side of things.

The whole system is connected by a T1 Tie line to my old Rolm system (a Plain 
T1...RBS, E&M Wink, etc..), and on the Rolm side, when this "sleep" issue crops 
up, my monitoring there either shows things like "RING-IN" or "ERROR", but no 
connection is actually made between the Rolm and Zaptel, so it's like the Rolm 
is getting a signal down the line, but not a signal it likes.  I see the Wink 
indicators, so I doubt it's the signalling or anything.  On the Asterisk side, 
it just says "Everyone is busy/congested at this time (1:0/0/1)" almost 
immediately after Dialing out, and then plays the no-service message.

But once you make a couple of call attempts on the line after a few days of 
inactivity, either fax calls via HylaFax or via an IAX softphone I setup for 
testing, it all just starts working again, which really baffles me.  Lately, I 
also found shutting down Asterisk and restarting zaptel seems to work too.  
It's kinda like getting out of bed after taking a long napalmost like the 
system is groggy and so poking it with a stick (repeated call attempts) or hit 
it with a bucket of water (restarting Asterisk/Zaptel) wake it up 
(unfortunately, my systems don't drink coffee).

Thoughts perchance?  Asterisk version is 1.4.18.1, and Zaptel is 1.4.9.2.


Thanks!,

--Josh

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[asterisk-users] Config file for 'make menuselect' available?

[Joshua Kinard]

Hi all,

Was curious to if for the 'make menuselect' command, there's a config file 
hiding someplace that lets me quickly move a configuration to a new source tree 
(much like .config in the kernel trees).  I looked around after running 
menuselect and compiling, but none of the files stood out as config files, so I 
thought I'd put the question to the list.

Thanks!,

--Josh

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Re: [asterisk-users] AsteriskNOW and IE

[Dean Collins]

Lol and Asterisk NOW isn't a commercial product - The list nazi's are
out today. 

Personally I'm ok with people announcing commercial products once and
once only on the user list and then taking it over to the biz list after
that as not everyone is a subscriber to the biz list.

Anyway - looks like a cool product. Good luck with it, will be
interesting to see how this goes for you especially as Voiceroute is
also out their competing with their druid gui they just announced a few
weeks ago as well.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Tilghman Lesher
> Sent: Thursday, 3 April 2008 12:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] AsteriskNOW and IE
> 
> On Thursday 03 April 2008 10:56:30 John Signorello wrote:
> > CogoBlue is currently only available on ISPBX's line of PBX
appliances.
> > You can check out the hardware specs on
> > http://ispbx.com/product_matrix.shtml
> 
> Now that you've announced your product, could you please move this
> discussion to the -biz list?  As specified in the title of this list,
this
> list is about NON-commercial discussion.
> 
> --
> Tilghman
> 
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Re: [asterisk-users] ztdummy

[Tzafrir Cohen]

On Thu, Apr 03, 2008 at 04:45:27PM +, Tony Mountifield wrote:
> In article <[EMAIL PROTECTED]>,
> Jerry Geis <[EMAIL PROTECTED]> wrote:
> > 
> > OK If I modprobe ztdummy then the RTC does start incrementing... to like 
> > 9250. then stops...
> > 
> > What now?
> 
> Sounds like the module got auto-unloaded due to not being in use.
> 
> I have found the most reliable way to invoke zaptel/ztdummy is using the
> proper init script:
> 
> 1. In your zaptel source directory, do "make config". That will create
> /etc/rc.d/init.d/zaptel and the rcX.d links to it.
> 2. Likeewise, in your asterisk source directory, also do "make config"
> if you haven't already.  That will create /etc/rc.d/init.d/asterisk and
> the rcX.d links.

All this machineray is not really required. All you need is to modprobe
ztdummy somewhere in the boot pcess . IIRC it can be even after asterisk
starts.

No real need to load any otyher modules. No need to run ztcfg. No need
to run anything else from the zaptel init.d script.

> 3. Do "chkconfig zaptel on" and "chkconfig asterisk on" just to be sure.
> 4. Do "service zaptel start". That will load the modules and do any
> configuration necessary. Try that and then watch /proc/interrupts.
> 5. Also do "service asterisk start".
> 6. When the system is booted up in future, zaptel and asterisk should
> automatically start.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances

[Tony Mountifield]

Matt,

I'm sure I won't be the only one to point out that your posts belongs in
asterisk-biz, not asterisk-users.

In article <[EMAIL PROTECTED]>,
Matt Signorello <[EMAIL PROTECTED]> wrote:
> [...]
> however because some dealers having problems with Asterisk NOW

If that is the justification for bringing your announcement forward,
then...

> COGOBLUE  is a true "Drag and Drop"
> Asterisk GUI available *on a series of ISPBX appliances*

... doesn't really help people who might (you perceive) be looking for
an alternative to AsteriskNOW, unless they discard their existing
hardware and buy yours.

It bothers me that a thread I started about a technical issue has been
used as a pretext for a commercial promotion that doesn't really address
the issues at hand.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] ztdummy

[Jerry Geis]

>
> ounds like the module got auto-unloaded due to not being in use.
>
> I have found the most reliable way to invoke zaptel/ztdummy is using the
> proper init script:
>
> 1. In your zaptel source directory, do "make config". That will create
> /etc/rc.d/init.d/zaptel and the rcX.d links to it.
> 2. Likeewise, in your asterisk source directory, also do "make config"
> if you haven't already.  That will create /etc/rc.d/init.d/asterisk and
> the rcX.d links.
> 3. Do "chkconfig zaptel on" and "chkconfig asterisk on" just to be sure.
> 4. Do "service zaptel start". That will load the modules and do any
> configuration necessary. Try that and then watch /proc/interrupts.
> 5. Also do "service asterisk start".
> 6. When the system is booted up in future, zaptel and asterisk should
> automatically start.
>
>   
Tony,

This is what  I do. In this case I also editied /etc/sysconfig/zaptel 
and removed (commented) all modules except ztdummy.

Jerry

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Re: [asterisk-users] AsteriskNOW and IE

[Tilghman Lesher]

On Thursday 03 April 2008 10:56:30 John Signorello wrote:
> CogoBlue is currently only available on ISPBX's line of PBX appliances.
> You can check out the hardware specs on
> http://ispbx.com/product_matrix.shtml

Now that you've announced your product, could you please move this
discussion to the -biz list?  As specified in the title of this list, this
list is about NON-commercial discussion.

-- 
Tilghman

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Re: [asterisk-users] ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> 
> OK If I modprobe ztdummy then the RTC does start incrementing... to like 
> 9250. then stops...
> 
> What now?

Sounds like the module got auto-unloaded due to not being in use.

I have found the most reliable way to invoke zaptel/ztdummy is using the
proper init script:

1. In your zaptel source directory, do "make config". That will create
/etc/rc.d/init.d/zaptel and the rcX.d links to it.
2. Likeewise, in your asterisk source directory, also do "make config"
if you haven't already.  That will create /etc/rc.d/init.d/asterisk and
the rcX.d links.
3. Do "chkconfig zaptel on" and "chkconfig asterisk on" just to be sure.
4. Do "service zaptel start". That will load the modules and do any
configuration necessary. Try that and then watch /proc/interrupts.
5. Also do "service asterisk start".
6. When the system is booted up in future, zaptel and asterisk should
automatically start.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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[asterisk-users] ISPBX Announces COGOBLUE Interface and PBX Appliances

[Matt Signorello]

Hi Everyone,

My name is Matt Signorello and I'm responsible for wholesale dealers
sales here at ISPBX. (www.ispbx.com)

ISPBX is a New Jersey based systems developer marketing a series of
solid state Asterisk appliances since 2005. For the last year we've been
working on a new strategy for helping dealers install Asterisk
appliances into the end user customer base in a smarter more profitable
way. 

Officially our launch date isn't until next Wednesday the 9th of April,
however because some dealers having problems with Asterisk NOW we've
decided to announce today and the official press release is available
for download at
http://www.ispbx.com/press/ispbx_cogoblue_press_040208.pdf

ISPBX is proud to announce COGOBLUE the 3rd generation of Asterisk GUI
tools. COGOBLUE  is a true "Drag and Drop"
Asterisk GUI available on a series of ISPBX appliances for small to
medium businesses.

For the first time dealers installing these ip-pbx appliances can
quickly implement an entire PBX installation without the need for manual
config file editing. COGOBLUE allows you to configure your system just
by dragging and dropping visual icons on a display pane.

Trunks, Extensions, Voicemail, Call groups, Auto attendants, etc etc can
all be configured simply by dragging and dropping the icons into their
appropriate place. Using a visual display to implement a call flow is
just a smarter, faster (and more profitable way) to implement Asterisk
installations for your customers.

Also when you are hired at a later date to provide additional 'moves and
changes' support times are shorter and more profitable because
technicians don't need to re-learn how it was previously configured -
instantly you can see in a visual display the entire PBX configuration.

We've provided some screen shots and video install demo's for you here
http://www.cogoblue.com/tourcogo.shtml

There is also a more extensive manual available here
http://www.cogoblue.com/help

The COGOBLUE GUI is available on a series of Asterisk appliances for
various hardware and user requirements
http://www.ispbx.com/product_matrix.shtml

We look forward to bringing about a smarter more profitable way to
provide Asterisk installations and meet your end users requirements and
look forward to your inquiries.

If you have questions please feel free to email me at [EMAIL PROTECTED] or
feel free to give me a call directly.

Thanks,

Matt Signorello
Managing Partner
ISPBX LLC
Direct: 973-841-2060



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Re: [asterisk-users] ztdummy

[Jerry Geis]

>
> Hi Jerry,
>
> Is that with ztdummy loaded or not? By default, Linux doesn't have 
> anything
> that uses the RTC interrupt, so without ztdummy it will usually stay 
> at 1.
>
> Once ztdummy and zaptel are loaded, then you should see it incrementing.
> If not, that suggests a problem.
>
> I have just installed ztdummy on a new system running 2.6.18-53.1.6.el5,
> and it is incrementing fine. I didn't realise there was a newer kernel 
> out;
> I'll have to update and try again.
>
> I didn't know the "watch" command - that's cool!
>
>   
Tony,

OK If I modprobe ztdummy then the RTC does start incrementing... to like 
9250. then stops...

What now?

Jerry

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[asterisk-users] fax detection on sip trunk

[ronald ramos]

Hi,

Is it possible for me to detect fax on a sip trunk?

my provider has a fax service that can send/receive
fax.

is it possible that i use a that trunk as a telefax?
meaning i will try to detect if it's a fax, if it is i
will forward it to an extension that can handle fax if
not will forward it elsewhere.

thank you

regards

ron


  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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Re: [asterisk-users] Wait for dialtone feature on FXO device

[Tilghman Lesher]

On Thursday 03 April 2008 10:37:32 John Novack wrote:
> WOW! Is this LONG overdue.
> Why this wasn't done initially is beyond me
> It has caused so many troubles and questions and posts from folks who
> expected Asterisk to at least have a feature that has been in a dial up
> modem for 10+ years.

Welcome to open source.  If you'd like to see a feature, might I suggest
writing it, and contributing it back?

-- 
Tilghman

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Re: [asterisk-users] ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Tony Mountifield <[EMAIL PROTECTED]> wrote:
> 
> I have just installed ztdummy on a new system running 2.6.18-53.1.6.el5,
> and it is incrementing fine. I didn't realise there was a newer kernel out;
> I'll have to update and try again.

Just updated to 2.6.18-53.1.14.el5 (i386), re-installed zaptel and ztdummy,
and the RTC interrupt counter is incrementing just fine.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] AsteriskNOW and IE

[John Signorello]

Dean:

CogoBlue is currently only available on ISPBX's line of PBX appliances.
You can check out the hardware specs on 
http://ispbx.com/product_matrix.shtml


We are growing our distribution channel and are actively looking
for qualified dealers to join us. Potential dealers wanting more 
information should drop

me an email at [EMAIL PROTECTED]

Our entry level PBX, Model 500, has a MSRP under $1200
We have an aggressive dealer discount program.

Let's be frank, there is a lot of nice hardware out there. But, what 
sets us apart
from the pack is CogoBlue. We took a radically different direction in 
providing a

rich, visual model for configuring your PBX.

You really have to see it to appreciate it.

We assembled a series of flash movies that show CogoBlue in action:

http://cogoblue.com/supportcogo_help.shtml

regards,

John Signorello
[EMAIL PROTECTED]
cogoblue.com
ispbx.com
(866) GO ISPBX











Dean Collins wrote:

Hi John,

I think my history is well documented within the asterisk community that
moving Asterisk out of the geek zone and into the mainstream business
space is good for everyone.

It's good for customers, and it's good for programmers looking for
funding for the next generation of Asterisk tools and applications.

Quick question though - whose boxes do you install on and is CogoBlue
available as a standalone application and how do dealers get involved
(and also what price points)?

 


Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 



  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John Signorello
Sent: Thursday, 3 April 2008 10:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AsteriskNOW and IE

To the Asterisk community,

Well this is as good a time to break cover as any, officially our


launch
  

date isn't until next Wednesday the 9th of April, but based on this
discussion I feel that we have to comment.

We are going to announce CogoBlue, the 3rd generation of Asterisk GUI


tools
  

 (well I guess we are announcing it now :)

www.CogoBlue.com is a true "Drag and Drop" Asterisk gui available on a
family of ISPBX appliances for small to medium businesses.

For the first time people installing these ip-pbx appliances can
quickly implement an entire pbx installation without the need for
manual conf file editing.

Using a visual display to implement call flow and processing is just a


smarter,
  

faster (and more profitable) way to implement Asterisk installations.

With regards to ie7. The asterisk community has evolved from a clique


of
  

linux guru's hand editing .conf files into a mainstream business tool.

Now while - everyone loves to prove how macho they are by coding in Vi
using elaborate call routing code routines from memoryour


customers
  

aren't that market any more.

Ie7 has 22% of the browser market (and probably higher in the


commercial
  

non-consumer space).

It is no longer an option to tell 1/5th of your customers, that to use


our
  

appliance you have to change your browser (which is why Kevin from
Digium jumped in so quickly to explain that a new groud up version is
being built right now to support ie7).

As the Asterisk community becomes more mature, with more and more
mainstream customers getting involved with this fantastic technology,
it's time to realize not everyone wants to climb a steep learning


curve to use our
  

tools.

We can still code our own home systems on beta code running all the
latest test code but we also have to understand that real business is
about stability and efficiency and being able to work within your
client's requirements..even if they run ie7 :)


Regards,
John Signorello
[EMAIL PROTECTED]
ispbx.com
(866) GO ISPBX


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Re: [asterisk-users] Analog modem as phone

[Jay R. Ashworth]

On Wed, Apr 02, 2008 at 02:40:49PM -0600, Greg Woods wrote:
> On Wed, 2008-04-02 at 21:18 +0200, Ronny Forberger wrote:
> > I want to use a analog V.92 modem to make outgoing (and possibly)  
> > incoming phone call through a standard analog phone line.
> 
> When I asked this question, I was basically told that it isn't possible.
> The problem is along the lines that the modem uses many more wavelengths
> and more bandwidth than a regular phone does, so this won't work through
> the card. I have found that I can send outgoing faxes, and incoming
> faxes redirected to this modem also work, but I have to patch the modem
> through directly to the wall plate in order to be able to make dialup
> connections.

Unless I misread the OP and the followups, Greg, Ronny isn't trying to
use a modem as an FXS device to talk through his Asterisk box, he wants
to use it as a one-channel FXO interface to let the Asterisk box talk
to the PSTN.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Joseph Stalin)

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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Steve Totaro]

I am sure they have until you for whatever business reason, you needed
to move 24 or 48 phones to a distant campus.

There is very little flexibility with T1 Adit solution.  Yes it works
well, no it is not flexible, no you you will never even close to what
the Quintum Tenor AX offers as far as features.

I think using SIP (or whatever protocol) over a voice T1 is kind of
the whole point of this "VoIP Revolution".

Thanks,
Steve Totaro

On Thu, Apr 3, 2008 at 10:18 AM, Darren Wright <[EMAIL PROTECTED]> wrote:
> I've used Adit600's almost exclusively for my installs.   All have worked 
> great for me.
>
>  -D
>
>
>  
>
>  From: [EMAIL PROTECTED] on behalf of Steve Totaro
>  Sent: Thu 4/3/2008 10:01 AM
>  To: Asterisk Users Mailing List - Non-Commercial Discussion
>  Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks.
>
>
>
>
>  Just Google Quintum Tenor AX.  Well worth the money.
>
>  Thanks,
>  Steve Totaro
>
>  On Mon, Mar 31, 2008 at 10:03 PM, Al lists <[EMAIL PROTECTED]> wrote:
>  > Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
>  >  Steve, what are my options for SIP to fxs?
>  >  thank you!
>  >
>  >
>  >
>  >  On 3/31/08, Doug Lytle <[EMAIL PROTECTED]> wrote:
>  >  > Don Pobanz wrote:
>  >  > > Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
>  >  > >
>  >  > >>
>  >  > >
>  >  > > This does not sound right. If it is 2 PRIs then it should be 46 
> channels
>  >  > >
>  >  > >
>  >  >
>  >  > I may have the terminology incorrect. I don't have a D channel, so I
>  >  > guess this would be called a T1 then?
>  >  >
>  >  > Doug
>  >  >
>  >  >
>  >  > --
>  >  > Ben Franklin quote:
>  >  >
>  >  > "Those who would give up Essential Liberty to purchase a little 
> Temporary
>  >  > Safety, deserve neither Liberty nor Safety."
>  >  >
>  >  >
>  >  > ___
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> >  >
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>  >  >
>  >
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> >
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>
>
>  This message was sent from D2 Technology, INC.
>
>
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Re: [asterisk-users] ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> > >/ uname -a shows x86_64 and Centos 5.1,  2.6.18-53.1.14.el5
> > /
> > You can try zttest, although I'd bet it will hang. See what's going  
> > to the console (or use dmesg.) If it's a lot of rtc errors, then  
> > you'll likely need to upgrade your kernel to at least 2.6.23.11. That  
> > worked for me.
> >
> >   
> You are correct zttest hangs.
> dmesg doesnt show anything, ztdummy loads fine. lsmod shows it.
> I see no rtc errors in dmesg either.
> 
> I have tried 2.6.24.4 and same situation. rtc just has 1 in 
> /proc/interrupts.
> 
> What now?

Is this system used for other things, or could you try installing a
32-bit (i386) version of CentOS and trying again?  64-bit vs 32-bit
is the only think I can think of at the moment.

Also, what do you get from: fgrep RTC /var/log/messages

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Norman Franke <[EMAIL PROTECTED]> wrote:
> On Apr 3, 2008, at 10:32 AM, [EMAIL PROTECTED]  
> wrote:
> 
> > uname -a shows x86_64 and Centos 5.1,  2.6.18-53.1.14.el5
> 
> You can try zttest, although I'd bet it will hang. See what's going  
> to the console (or use dmesg.) If it's a lot of rtc errors, then  
> you'll likely need to upgrade your kernel to at least 2.6.23.11. That  
> worked for me.

I'd be surprised if that is the solution - I have been using ztdummy with
the RTC hook since 2.6.9 with no problems, and again on 2.6.18-53.1.6.el5

Unless it's a 64-bit issue - I've only ever used 32-bit.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Wait for dialtone feature on FXO device

[John Novack]

WOW! Is this LONG overdue.
Why this wasn't done initially is beyond me
It has caused so many troubles and questions and posts from folks who 
expected Asterisk to at least have a feature that has been in a dial up 
modem for 10+ years.

Great job!
Many thanks

John  Novack


Steve Davies wrote:
> Anyone interested in this feature? I have a version 0.1 patch, which
> is currently against 1.2.25-bristuffed, but which should port
> trivially to almost any version. I am away until Tuesday 8th April,
> but if there is enough interest, I will open a "new-feature" ticket
> and upload the patch to the bugtracker so that more capable
> programmers can laugh at it ;-)
>
> It should work reasonably on North-American and UK systems, which seem
> to use the same dialtone frequencies.
>
> Shout if interested.
> Steve
>
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Re: [asterisk-users] ztdummy

[Jerry Geis]

>/ uname -a shows x86_64 and Centos 5.1,  2.6.18-53.1.14.el5
/
You can try zttest, although I'd bet it will hang. See what's going  
to the console (or use dmesg.) If it's a lot of rtc errors, then  
you'll likely need to upgrade your kernel to at least 2.6.23.11. That  
worked for me.


  

You are correct zttest hangs.
dmesg doesnt show anything, ztdummy loads fine. lsmod shows it.
I see no rtc errors in dmesg either.

I have tried 2.6.24.4 and same situation. rtc just has 1 in 
/proc/interrupts.


What now?

Jerry
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Re: [asterisk-users] ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> > On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
> >
> > >/ Jerry, the first thing to check is "cat /proc/interrupts" and see if 
> > >there
> > />/ is an entry for rtc on IRQ 8. There should be, and the interrupt counts
> > />/ on there should be going up at approximately 1024 per second.
> > /
> > To see it better:
> >
> > watch -n1 -d cat /proc/interrupts
> >
> >   
> watch -n1 -d cat /proc/interrupts
> Every 1.0s: cat 
> /proc/interrupts  
>   
> 
> Thu Apr  3 10:29:19 2008
> 
>CPU0   CPU1
>   0:129  0XT-PIC-XTtimer
>   1: 337248  0XT-PIC-XTi8042
>   2:  0  0XT-PIC-XTcascade
>   3:  4  0XT-PIC-XT
>   4: 69  0XT-PIC-XT
>   5:  737837152  0XT-PIC-XTHDA Intel
>   7:385  0XT-PIC-XTparport0
>   8:  1  0XT-PIC-XTrtc
> 
> 
> It still remains as 1...

Hi Jerry,

Is that with ztdummy loaded or not? By default, Linux doesn't have anything
that uses the RTC interrupt, so without ztdummy it will usually stay at 1.

Once ztdummy and zaptel are loaded, then you should see it incrementing.
If not, that suggests a problem.

I have just installed ztdummy on a new system running 2.6.18-53.1.6.el5,
and it is incrementing fine. I didn't realise there was a newer kernel out;
I'll have to update and try again.

I didn't know the "watch" command - that's cool!

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] Call signalling on BT FeatureLine Compact (Sangoma A200)

[Paul Goodyear]

Update,

Still not sorted, I have checked some tools on the TrixBox and using
the wanrouter I was able to check the voltage on lines.


The three result are when there is no call active

[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 1
--- Voltage Status  (FXO,port 0) ---
VOLTAGE : 0 Volts

[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 2
--- Voltage Status  (FXO,port 1) ---
VOLTAGE : 49 Volts

[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 3
--- Voltage Status  (FXO,port 2) ---
VOLTAGE : 52 Volts


The following three are when a call is placed

[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 1
--- Voltage Status  (FXO,port 0) ---
VOLTAGE : 1 Volts

[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 2
--- Voltage Status  (FXO,port 1) ---
VOLTAGE : 7 Volts

[EMAIL PROTECTED] ~]# wanpipemon -i w1g1 -c astats -m 3
--- Voltage Status  (FXO,port 1) ---
VOLTAGE : 7 Volts



Line 1 shows no Volts when on the hook, but off the hook 1 Volt where
as Lines 2 and 3 (working fine) show the same results with 49/52Volts
on hook and 7Volts off hook.

Does this help anyone?

Thanks, PaulG.



On Thu, Mar 20, 2008 at 8:44 AM, David Quinton <[EMAIL PROTECTED]> wrote:
> On Wed, 19 Mar 2008 10:10:21 + (GMT), Gordon Henderson
>
> <[EMAIL PROTECTED]> wrote:
>
>
> >> I got free installation for Featureline Compact
>  >> on 3 yr contract.
>  >> So it saved me £££s!
>  >
>  >Intersting... But shouldn't you be using VoIP for your calls anyway...
>  >Then just one basic BT line, and a business-quality ADSL service, then you
>  >can bypass all that nasty horrible analogy echoy stuff :)
>
>  Problem is that we can only manage a measly 1.5 - 2Mb downstream here
>  and we've already got ISDN2 for incoming.
>
>  Outgoing are via my Trixbox which tries the sequence:
>  ENUM
>  SIP (we use Orbtalk)
>  ISDN
>
>
>
>
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Re: [asterisk-users] Web page to show online extensions?

[Earl Terwilliger]

You might try this:

http://www.micpc.com/eventmonitor/

It is php, and you can easily disable what you don't want..

earl

On Thursday 03 April 2008 09:41:52 am Vincent wrote:
> Hello
>
> Has someone written a web page (preferably PHP) that simply shows what
> extensions are currently online?
>
> Thank you.
>
>
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Re: [asterisk-users] AsteriskNOW and IE

[Dean Collins]

Hi John,

I think my history is well documented within the asterisk community that
moving Asterisk out of the geek zone and into the mainstream business
space is good for everyone.

It's good for customers, and it's good for programmers looking for
funding for the next generation of Asterisk tools and applications.

Quick question though - whose boxes do you install on and is CogoBlue
available as a standalone application and how do dealers get involved
(and also what price points)?

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of John Signorello
> Sent: Thursday, 3 April 2008 10:33 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] AsteriskNOW and IE
> 
> To the Asterisk community,
> 
> Well this is as good a time to break cover as any, officially our
launch
> date isn't until next Wednesday the 9th of April, but based on this
> discussion I feel that we have to comment.
> 
> We are going to announce CogoBlue, the 3rd generation of Asterisk GUI
tools
>  (well I guess we are announcing it now :)
> 
> www.CogoBlue.com is a true "Drag and Drop" Asterisk gui available on a
> family of ISPBX appliances for small to medium businesses.
> 
> For the first time people installing these ip-pbx appliances can
> quickly implement an entire pbx installation without the need for
> manual conf file editing.
> 
> Using a visual display to implement call flow and processing is just a
smarter,
> faster (and more profitable) way to implement Asterisk installations.
> 
> With regards to ie7. The asterisk community has evolved from a clique
of
> linux guru's hand editing .conf files into a mainstream business tool.
> 
> Now while - everyone loves to prove how macho they are by coding in Vi
> using elaborate call routing code routines from memoryour
customers
> aren't that market any more.
> 
> Ie7 has 22% of the browser market (and probably higher in the
commercial
> non-consumer space).
> 
> It is no longer an option to tell 1/5th of your customers, that to use
our
> appliance you have to change your browser (which is why Kevin from
> Digium jumped in so quickly to explain that a new groud up version is
> being built right now to support ie7).
> 
> As the Asterisk community becomes more mature, with more and more
> mainstream customers getting involved with this fantastic technology,
> it's time to realize not everyone wants to climb a steep learning
curve to use our
> tools.
> 
> We can still code our own home systems on beta code running all the
> latest test code but we also have to understand that real business is
> about stability and efficiency and being able to work within your
> client's requirements..even if they run ie7 :)
> 
> 
> Regards,
> John Signorello
> [EMAIL PROTECTED]
> ispbx.com
> (866) GO ISPBX
> 
> 
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Re: [asterisk-users] AsteriskNOW and IE

[Roderick A. Anderson]

Tzafrir Cohen wrote:
> On Thu, Apr 03, 2008 at 06:06:19AM -0700, Roderick A. Anderson wrote:
> 
>> IE was still on the desktop because they had to support a lot 
>> of customers that used IE but for in-house stuff it slowly became a 
>> Firefox place.
> 
> That's no excuse.
> 
> That's what IETab's for.

This was two+plus years ago.  As I remember IE didn't get usable TABs 
until a little over a year ago.  Probably my bad memory as I only do 
Windows when under duress.  :-)


Rod
-- 
> 


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Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Drew Gibson]

Yes Dean,

we do use Asterisk at OANDA. We've been running our office on it for 2 
years now and the call centre's 1st Asterisk anniversary was April 1!

Glad you like our site, we've just launched FXGame Mobile 
(http://fxlabs.oanda.com) for the gadget lover and FXGlobal Transfer for 
ex-pat Aussies to send their money home. :-)

Perhaps I should have added a smiley for my fellow North Americans but I 
think my analysis is valid. Perhaps the interesting points is that % of 
Skype users seems to be fairly uniform across cultures around the world  
although perhaps a little higher in Europe (a.m. usage while North 
Americans are still sleeping). I suspect that this is due to the call 
billing structure in Europe. They make the North American telcos look 
positively philanthropic.

regards,

Drew


Dean Collins wrote:
> Wow Drew, I had no idea someone from Oanda was a subscriber to the
> Asterisk list (and therefore an asterisk user company?).
>
> Just wanted to say you guys run a fantastic sight and I've been a long
> time user for at least the last 2 years.
>
>
>
>
> Now for the irony part of your email. I found it interesting with
> regards to global penetration and world population.
>
> I'd also be interested in seeing some comparisons where this goes
> against the trend.
>
> Eg would love to see a transaction per hour chart for Oanda on a global
> basis (though I don't know what your customer spread is).
>
> I've already sent this email to the creator of twitter vision to see if
> he can do something similar against tweets per hour on a rolling basis.
>
> I'd also love to see something truly global like SMS use which is pretty
> much a global application. Something like number of SMS's sent per hour
> across the various countries over a 24 hour period would be hugely
> interesting (well to me anyway).
>
>  
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED] 
> +1-212-203-4357
> +61-2-9016-5642 (Sydney in-dial). 
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Drew Gibson
>> Sent: Thursday, 3 April 2008 9:38 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling
>> 
> Analytics
>   
>> Dean Collins wrote:
>> 
>>>   
> http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analyt
> ics.html
>   
>>> >>   
>> vgIjvI/Af0/PgE_8gFqrY8/s1600-
>> h/World%2Bpopulation%2Bawake.png>
>> 
>>> Totally stumbled across this really interesting post
>>> http://skypejournal.com/blog/2008/04/world_online_or_asleep.html
>>>
>>>   
>> So, it seems that people make more calls on Skype when they are not
>> sleeping and that people who don't have computers don't make so many
>> Skype calls, even when they are awake.
>>
>> Fascinating!
>>
>> regards,
>>
>> Drew
>> 

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Tzafrir Cohen]

On Thu, Apr 03, 2008 at 10:18:14AM -0400, Dean Collins wrote:
> Wow Drew, I had no idea someone from Oanda was a subscriber to the
> Asterisk list (and therefore an asterisk user company?).
> 
> Just wanted to say you guys run a fantastic sight and I've been a long
> time user for at least the last 2 years.
> 
> 
> 
> 
> Now for the irony part of your email. I found it interesting with
> regards to global penetration and world population.
> 
> I'd also be interested in seeing some comparisons where this goes
> against the trend.
> 
> Eg would love to see a transaction per hour chart for Oanda on a global
> basis (though I don't know what your customer spread is).
> 
> I've already sent this email to the creator of twitter vision to see if
> he can do something similar against tweets per hour on a rolling basis.
> 
> I'd also love to see something truly global like SMS use which is pretty
> much a global application. Something like number of SMS's sent per hour
> across the various countries over a 24 hour period would be hugely
> interesting (well to me anyway).

Here's another insight on actions of some people:

http://ibot.rikers.org/stats/asterisk.html.gz

Yeah, I know, it's not web 2.0, so it's irrelevant. But still.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Wait for dialtone feature on FXO device

[Steve Davies]

Anyone interested in this feature? I have a version 0.1 patch, which
is currently against 1.2.25-bristuffed, but which should port
trivially to almost any version. I am away until Tuesday 8th April,
but if there is enough interest, I will open a "new-feature" ticket
and upload the patch to the bugtracker so that more capable
programmers can laugh at it ;-)

It should work reasonably on North-American and UK systems, which seem
to use the same dialtone frequencies.

Shout if interested.
Steve

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Re: [asterisk-users] ztdummy

[Norman Franke]

On Apr 3, 2008, at 10:32 AM, [EMAIL PROTECTED]  
wrote:



uname -a shows x86_64 and Centos 5.1,  2.6.18-53.1.14.el5


You can try zttest, although I'd bet it will hang. See what's going  
to the console (or use dmesg.) If it's a lot of rtc errors, then  
you'll likely need to upgrade your kernel to at least 2.6.23.11. That  
worked for me.


Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

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Re: [asterisk-users] Send DTMF digit every 15 seconds during a call

[Alexander Lopez]

Use call file to call out to the Alarm Panel and them put it in a
context that would do this:

[alarm-keepup]
exten => s,1,Answer
exten => s,2,SendDTMF(1)
exten => s,3,Wait(15)
exten => s,4,Goto(s,2)

You did not specify if you needed to do anything other than send the
digit to the alarm panel. If you are waiting for an answer from the
panel or you need to bridge the panel to something else then this WILL
NOT work.



> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Thursday, April 03, 2008 10:13 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Send DTMF digit every 15 seconds during a
call
> 
> 
> I am trying to send a DTMF digit automatically every 15 seconds to
keep a
> call connected to an alarm panel.  I tried using the dial command L
and
> recording a dtmf tone for the beep, but obviously that didn't work.
Does
> anyone have a suggestion for merging the L option and the sendDTMF or
the
> D
> option?  Any other suggestions would be appreciated!
> 
> Thanks!
> Paul Gentilini
> 
> 
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Re: [asterisk-users] AsteriskNOW and IE

[John Signorello]

To the Asterisk community,

Well this is as good a time to break cover as any, officially our launch
date isn't until next Wednesday the 9th of April, but based on this
discussion I feel that we have to comment.

We are going to announce CogoBlue, the 3rd generation of Asterisk GUI tools
 (well I guess we are announcing it now :)

www.CogoBlue.com is a true "Drag and Drop" Asterisk gui available on a
family of ISPBX appliances for small to medium businesses.

For the first time people installing these ip-pbx appliances can
quickly implement an entire pbx installation without the need for
manual conf file editing. 

Using a visual display to implement call flow and processing is just a smarter,
faster (and more profitable) way to implement Asterisk installations.

With regards to ie7. The asterisk community has evolved from a clique of
linux guru's hand editing .conf files into a mainstream business tool.

Now while - everyone loves to prove how macho they are by coding in Vi
using elaborate call routing code routines from memoryour customers
aren't that market any more.

Ie7 has 22% of the browser market (and probably higher in the commercial
non-consumer space).

It is no longer an option to tell 1/5th of your customers, that to use our
appliance you have to change your browser (which is why Kevin from
Digium jumped in so quickly to explain that a new groud up version is
being built right now to support ie7).

As the Asterisk community becomes more mature, with more and more
mainstream customers getting involved with this fantastic technology, 
it's time to realize not everyone wants to climb a steep learning curve to use 
our tools.

We can still code our own home systems on beta code running all the
latest test code but we also have to understand that real business is
about stability and efficiency and being able to work within your
client's requirements..even if they run ie7 :)


Regards,
John Signorello
[EMAIL PROTECTED]
ispbx.com
(866) GO ISPBX


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Re: [asterisk-users] ztdummy

[Jerry Geis]

On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:

>/ Jerry, the first thing to check is "cat /proc/interrupts" and see if there
/>/ is an entry for rtc on IRQ 8. There should be, and the interrupt counts
/>/ on there should be going up at approximately 1024 per second.
/
To see it better:

watch -n1 -d cat /proc/interrupts

  

watch -n1 -d cat /proc/interrupts
Every 1.0s: cat 
/proc/interrupts
Thu Apr  3 10:29:19 2008


  CPU0   CPU1
 0:129  0XT-PIC-XTtimer
 1: 337248  0XT-PIC-XTi8042
 2:  0  0XT-PIC-XTcascade
 3:  4  0XT-PIC-XT
 4: 69  0XT-PIC-XT
 5:  737837152  0XT-PIC-XTHDA Intel
 7:385  0XT-PIC-XTparport0
 8:  1  0XT-PIC-XTrtc


It still remains as 1...

Jerry

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Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Dean Collins]

Wow Drew, I had no idea someone from Oanda was a subscriber to the
Asterisk list (and therefore an asterisk user company?).

Just wanted to say you guys run a fantastic sight and I've been a long
time user for at least the last 2 years.




Now for the irony part of your email. I found it interesting with
regards to global penetration and world population.

I'd also be interested in seeing some comparisons where this goes
against the trend.

Eg would love to see a transaction per hour chart for Oanda on a global
basis (though I don't know what your customer spread is).

I've already sent this email to the creator of twitter vision to see if
he can do something similar against tweets per hour on a rolling basis.

I'd also love to see something truly global like SMS use which is pretty
much a global application. Something like number of SMS's sent per hour
across the various countries over a 24 hour period would be hugely
interesting (well to me anyway).

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Drew Gibson
> Sent: Thursday, 3 April 2008 9:38 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling
Analytics
> 
> Dean Collins wrote:
> >
> >
http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analyt
ics.html
> >
> >
> >  vgIjvI/Af0/PgE_8gFqrY8/s1600-
> h/World%2Bpopulation%2Bawake.png>
> >
> >
> > Totally stumbled across this really interesting post
> > http://skypejournal.com/blog/2008/04/world_online_or_asleep.html
> >
> 
> So, it seems that people make more calls on Skype when they are not
> sleeping and that people who don't have computers don't make so many
> Skype calls, even when they are awake.
> 
> Fascinating!
> 
> regards,
> 
> Drew
> 
> --
> Drew Gibson
> 
> Systems Administrator
> OANDA Corporation
> www.oanda.com
> 
> 
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Re: [asterisk-users] Web page to show online extensions?

[Jonn Taylor]

Its called Flash Operator Panel or FOP. It is install with freepbx, but 
I think you can use it as a standalone app.

Jonn

Vincent wrote:
> Hello
>
> Has someone written a web page (preferably PHP) that simply shows what
> extensions are currently online?
>
> Thank you.
>
>
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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Darren Wright]

I've used Adit600's almost exclusively for my installs.   All have worked great 
for me.
 
-D
 



From: [EMAIL PROTECTED] on behalf of Steve Totaro
Sent: Thu 4/3/2008 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Need some input for Quad T1 and channel banks.



Just Google Quintum Tenor AX.  Well worth the money.

Thanks,
Steve Totaro

On Mon, Mar 31, 2008 at 10:03 PM, Al lists <[EMAIL PROTECTED]> wrote:
> Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
>  Steve, what are my options for SIP to fxs?
>  thank you!
>
>
>
>  On 3/31/08, Doug Lytle <[EMAIL PROTECTED]> wrote:
>  > Don Pobanz wrote:
>  > > Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
>  > >
>  > >>
>  > >
>  > > This does not sound right. If it is 2 PRIs then it should be 46 channels
>  > >
>  > >
>  >
>  > I may have the terminology incorrect. I don't have a D channel, so I
>  > guess this would be called a T1 then?
>  >
>  > Doug
>  >
>  >
>  > --
>  > Ben Franklin quote:
>  >
>  > "Those who would give up Essential Liberty to purchase a little Temporary
>  > Safety, deserve neither Liberty nor Safety."
>  >
>  >
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This message was sent from D2 Technology, INC.

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Re: [asterisk-users] Listening on conversations for training?

[Alexander Lopez]

Look at the ChanSpy Application. It would be the easiest to implement
and it also allows the trainee to speak to the support person without
the customer knowing.

You can also use on-demand recording or simply record ALL calls
(legality and disclosure to calling parties are outside the scope of
this document) then the trainee(s) can just access the calls that were
recorded. That may easier than stopping a support person to have them
call a trainee to 'jump-in'

Alex
 

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Vincent
> Sent: Thursday, April 03, 2008 9:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Listening on conversations for training?
> 
> Hello
> 
>   I assume it's possible to do this with Asterisk: To train a new
> worker who works remotely, I'd like to have him listen in on support
> calls so that he gets to learn the kind of problems that come in, and
> how they're solved.
> When a call comes in and the support person thinks it's worthy to have
> the trainee be part of it, he will ring the trainee so he can join the
> call.
> 
> From what I read, there seems to be two ways to do this:
> - either create a conference call, in which case the customer knows
> that a third party is part of the call
> - or have the trainee listen in on the conversation, unbeknownst to
> the customer
> 
> Does someone use Asterisk for this purpose, and could tell me what the
> best solution is, and how to set things up?
> 
> Thank you.
> 
> 
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Re: [asterisk-users] Web page to show online extensions?

[Dean Collins]

You mean like FOP?
http://www.asternic.org 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Vincent
> Sent: Thursday, 3 April 2008 10:12 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Web page to show online extensions?
> 
> Hello
> 
> Has someone written a web page (preferably PHP) that simply shows what
> extensions are currently online?
> 
> Thank you.
> 
> 
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[asterisk-users] Send DTMF digit every 15 seconds during a call

[PGentilini]

I am trying to send a DTMF digit automatically every 15 seconds to keep a
call connected to an alarm panel.  I tried using the dial command L and
recording a dtmf tone for the beep, but obviously that didn't work.  Does
anyone have a suggestion for merging the L option and the sendDTMF or the D
option?  Any other suggestions would be appreciated!

Thanks!
Paul Gentilini


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[asterisk-users] Web page to show online extensions?

[Vincent]

Hello

Has someone written a web page (preferably PHP) that simply shows what
extensions are currently online?

Thank you.


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Re: [asterisk-users] ztdummy

[Tzafrir Cohen]

On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:

> Jerry, the first thing to check is "cat /proc/interrupts" and see if there
> is an entry for rtc on IRQ 8. There should be, and the interrupt counts
> on there should be going up at approximately 1024 per second.

To see it better:

watch -n1 -d cat /proc/interrupts

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Steve Totaro]

Just Google Quintum Tenor AX.  Well worth the money.

Thanks,
Steve Totaro

On Mon, Mar 31, 2008 at 10:03 PM, Al lists <[EMAIL PROTECTED]> wrote:
> Im guessing T1cas not PRI,just because its giving 24 fxs per T1.
>  Steve, what are my options for SIP to fxs?
>  thank you!
>
>
>
>  On 3/31/08, Doug Lytle <[EMAIL PROTECTED]> wrote:
>  > Don Pobanz wrote:
>  > > Doug Lytle wrote on Monday, March 31, 2008 5:40 PM
>  > >
>  > >>
>  > >
>  > > This does not sound right. If it is 2 PRIs then it should be 46 channels
>  > >
>  > >
>  >
>  > I may have the terminology incorrect. I don't have a D channel, so I
>  > guess this would be called a T1 then?
>  >
>  > Doug
>  >
>  >
>  > --
>  > Ben Franklin quote:
>  >
>  > "Those who would give up Essential Liberty to purchase a little Temporary
>  > Safety, deserve neither Liberty nor Safety."
>  >
>  >
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Re: [asterisk-users] ztdummy

[Jerry Geis]

In article <47F4C604.1060301 at pagestation.com 
>,
Jerry Geis http://lists.digium.com/mailman/listinfo/asterisk-users>> wrote:
>/ What does it take to get ztdummy to work correctly?
/>/ 
/>/ I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 
/>/ 1.4.9.2

/>/ Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs.
/>/ Problem is playback() does not work. So then I stop zaptel, asterisk 
/>/ runs and playback() now
/>/ works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for 
/>/ that.
/>/ 
/>/ I am running Centos 5.1 with the latest kernel and fully updated.
/>/ 
/>/ How can I get ztdummy to work? There are no cards in the laptop.
/>/ I typically dont run asterisk on a laptop - but thats what the situation 
/>/ needed...

/
Jerry, the first thing to check is "cat /proc/interrupts" and see if there
is an entry for rtc on IRQ 8. There should be, and the interrupt counts
on there should be going up at approximately 1024 per second.

What kernel version are you using? (uname -a)

  

Tony,

/proc/interrupts shows:

   CPU0   CPU1
8:   1   0   IO-APIC-edge rtc

So the RTC is not incrementing.

uname -a shows x86_64 and Centos 5.1,  2.6.18-53.1.14.el5

Jerry

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Re: [asterisk-users] ztdummy

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> What does it take to get ztdummy to work correctly?
> 
> I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 
> 1.4.9.2
> Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs.
> Problem is playback() does not work. So then I stop zaptel, asterisk 
> runs and playback() now
> works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for 
> that.
> 
> I am running Centos 5.1 with the latest kernel and fully updated.
> 
> How can I get ztdummy to work? There are no cards in the laptop.
> I typically dont run asterisk on a laptop - but thats what the situation 
> needed...

Jerry, the first thing to check is "cat /proc/interrupts" and see if there
is an entry for rtc on IRQ 8. There should be, and the interrupt counts
on there should be going up at approximately 1024 per second.

What kernel version are you using? (uname -a)

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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Re: [asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Drew Gibson]

Dean Collins wrote:
>
>  
> http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analytics.html
>  
>
>
> 
>  
>
>
> Totally stumbled across this really interesting post 
> http://skypejournal.com/blog/2008/04/world_online_or_asleep.html
>

So, it seems that people make more calls on Skype when they are not 
sleeping and that people who don't have computers don't make so many 
Skype calls, even when they are awake.

Fascinating!

regards,

Drew

-- 
Drew Gibson

Systems Administrator
OANDA Corporation
www.oanda.com


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[asterisk-users] Listening on conversations for training?

[Vincent]

Hello

I assume it's possible to do this with Asterisk: To train a new
worker who works remotely, I'd like to have him listen in on support
calls so that he gets to learn the kind of problems that come in, and
how they're solved.
When a call comes in and the support person thinks it's worthy to have
the trainee be part of it, he will ring the trainee so he can join the
call.

>From what I read, there seems to be two ways to do this:
- either create a conference call, in which case the customer knows
that a third party is part of the call
- or have the trainee listen in on the conversation, unbeknownst to
the customer

Does someone use Asterisk for this purpose, and could tell me what the
best solution is, and how to set things up?

Thank you.


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Re: [asterisk-users] AsteriskNOW and IE

[Tzafrir Cohen]

On Thu, Apr 03, 2008 at 06:06:19AM -0700, Roderick A. Anderson wrote:

> IE was still on the desktop because they had to support a lot 
> of customers that used IE but for in-house stuff it slowly became a 
> Firefox place.

That's no excuse.

That's what IETab's for.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] AsteriskNOW and IE

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Roderick A. Anderson <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
> > a message at the top "Your browser is not supported by this version of 
> > GUI!",
> > and "We recommend using Firefox".
> > 
> > Does this mean that it is known NOT to work under IE7, or just that it is
> > insufficiently tested to be guaranteed?
> > 
> > It's easy to ask techies to use Firefox, but if we are trying to sell a
> > system to a customer who always uses IE, it is a bit of a negative point
> > to tell him, "oh, by the way, you have to download Firefox onto your PC,
> > because this doesn't work with IE."
> 
> I trick(?) used at a place I worked for was to install Firefox and make 
> a desktop shortcut to it for the GUI.
> 
> The interesting part was that once the users got using Firefox and 
> realized it did browsing better (mostly using tabs) they converted to 
> Firefox.  IE was still on the desktop because they had to support a lot 
> of customers that used IE but for in-house stuff it slowly became a 
> Firefox place.

I like it - a bit of social engineering!

Thanks to all for the responses on this thread, and to Kevin for the info
about there being an upcoming new GUI. I'll watch with interest - I've
just subscribed to asterisk-gui, so I assume info will be forthcoming
on there.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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